--- /dev/null
+Wim Taymans <wim.taymans@collabora.co.uk>
+Hyunjun Ko <zzoon.ko@samsung.com>
--- /dev/null
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--- /dev/null
+ GNU LIBRARY GENERAL PUBLIC LICENSE
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+ <signature of Ty Coon>, 1 April 1990
+ Ty Coon, President of Vice
+
+That's all there is to it!
--- /dev/null
+=== release 1.4.5 ===
+
+2014-12-18 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.4.5
+
+2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: unref srtp decoder when leaving bin
+ https://bugzilla.gnome.org/show_bug.cgi?id=739481
+
+=== release 1.4.4 ===
+
+2014-11-06 13:18:04 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.4.4
+
+2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: mikey memory leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=739383
+
+=== release 1.4.3 ===
+
+2014-09-24 12:51:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.4.3
+
+=== release 1.4.2 ===
+
+2014-09-19 15:13:37 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.4.2
+
+2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Make sure that sequence numbers are monotonic after pause
+ The sequence number is not monotonic for RTP packets after pause. The
+ reason is basepayloader generates a randon sequence number when the
+ pipeline goes from ready to pause. With this fix generation of sequence
+ number will be monotonic when going from pause to play request.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736017
+
+2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Protect saved clients watch with a mutex
+ Fixes a crash when close() is called while merging clients
+ in handle_tunnel(). In that case close() would destroy the
+ watch while it is still being used in handle_tunnel().
+ https://bugzilla.gnome.org/show_bug.cgi?id=735570
+
+=== release 1.4.1 ===
+
+2014-08-27 15:05:07 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.4.1
+
+2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Query position and stop time only on the RTP parts of the pipeline
+ The RTCP parts, in specific the RTCP udpsinks, are not flushed when
+ seeking and will always continue counting the time. This leads to
+ the NPT after a backwards seek to be something completely different
+ to the actual seek position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732644
+
+2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
+
+ * gst/rtsp-server/rtsp-media.c:
+ signals: Fix copy-pasto in target-state signal offset
+
+=== release 1.4.0 ===
+
+2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.4.0
+
+2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ media: correct misspelled words in description
+ https://bugzilla.gnome.org/show_bug.cgi?id=733244
+
+=== release 1.3.91 ===
+
+2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.91
+
+2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: update docs
+
+2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: implement client REMOVE filter
+
+2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: expose _close() method
+ Expose a previously internal close method to close the client
+ connection.
+
+2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session-pool: signal session-removed outside of the lock
+ Release the lock before emiting the session-removed signal.
+
+2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ filter: Release lock in filter functions
+ Release the object lock before calling the filter functions. We need to
+ keep a cookie to detect when the list changed during the filter
+ callback. We also keep a hashtable to make sure we only call the filter
+ function once for each object in case of concurrent modification.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
+
+2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: check if watch is set in handle_teardown()
+ The unit tests run without a watch
+
+2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/client.c:
+ client tests: send teardown to cleanup session
+
+2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ server tests: send teardown to cleanup session
+
+2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: keep ref to client for the session removed handler
+ This extra ref will be dropped when all client sessions have been
+ removed. A session is removed when a client sends teardown, closes its
+ endpoint of the TCP connection or the sessions expires.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * tests/check/gst/client.c:
+ client: manage media in session as a last step
+ Once we manage a media in a session, we can't unmanage it anymore
+ without destroying it. Therefore, first check everything before we
+ manage the media, otherwise if something is wrong we have no way to
+ unmanage the media.
+ If we created a new session and something went wrong, remove the session
+ again. Fixes a leak in the unit test.
+
+2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ examples: print 'stream ready at url' for mp4 and ogg example
+
+2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp: fix for MIKEY api change
+
+2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: free watch context only once
+ The watch context is freed when the source is destroyed. Avoids
+ a CRITICAL when we try to unref the context twice.
+
+2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix build
+
+2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: protect sessions with lock
+ Protect the list of sessions with the lock.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Client: keep a ref to the session
+ Don't just keep a weak ref to the session objects but use a hard ref. We
+ will be notified when a session is removed from the pool (expired) with
+ the new session-removed signal.
+ Don't automatically close the RTSP connection when all the sessions of
+ a client are removed, a client can continue to operate and it can create
+ a new session if it wants. If you want to remove the client from the
+ server, you have to use gst_rtsp_server_client_filter() now.
+ Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ session-pool: add session-removed signal
+ Add a signal to be notified when a session is removed from the pool.
+
+2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-server.h:
+ Make rtsp-server.h a single-include header, use it for G-I
+ https://bugzilla.gnome.org/show_bug.cgi?id=732411
+
+=== release 1.3.90 ===
+
+2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.90
+
+2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: crypto can be NULL
+
+2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ introspection: add missing allow-none annotations
+ https://bugzilla.gnome.org/show_bug.cgi?id=730952
+
+2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-token.c:
+ introspection: add (nullable) annotations to return values
+ https://bugzilla.gnome.org/show_bug.cgi?id=730952
+
+2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ gi: improve annotations
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
+
+2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ signals: use generic marshal function
+ Use the generic C marshal function.
+ Use more explicit type instead of G_TYPE_POINTER
+
+2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-context.h:
+ context: add type macro
+
+2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: hide key length defines
+ They don't have a namespace.
+
+2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.3.3 ===
+
+2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.3
+
+2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ mikey: add different key length parameters
+ Add encryption and authentication key length parameters to MIKEY. For
+ the encoders, the key lengths are obtained from the cipher and auth
+ algorithms set in the caps. For the decoders, they are obtained while
+ parsing the key management from the client.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
+
+2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
+
+ * tests/check/gst/stream.c:
+ stream tests: Make sure we get right multicast address from stream
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
+
+2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: ref the context until rtsp watch is alive
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
+
+2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Destroy the rtsp watch after connection close
+
+2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix confusing comment
+
+2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: Timeout in header.
+ Adding the possbilty to always have timout in header.
+ This is configurabe with setting "timeout-always-visible".
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
+
+2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.3.2 ===
+
+2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * common:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.2
+
+2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 211fa5f to 1f5d3c3
+
+2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: store TCP ports in transport
+ Store the TCP ports in the transport when we are doing RTSP over TCP.
+ This way, we can easily get to the ports from the transport.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
+
+2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: add signals for new RTP/RTCP encoders
+ New signals to allow the user to configure the dynamically created
+ encoders.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730228
+
+2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: Make suspend()/unsuspend() virtual
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
+
+2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix send-message signal marshaller
+ Use generic marshalling for the send-message signal. It has
+ two POINTER arguments, not just one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=729900
+
+2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/gst/media.c:
+ tests: add and remove pads only once
+ In this test we simulate a dynamic pad by watching the caps event.
+ Because of renegotiation in the base payloader now, this caps is sent
+ multiple times but we can only deal with 1 invocation, use a variable to
+ only 'add and remove' the pad once.
+
+2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: add unit test for correct handling of Require headers
+ https://bugzilla.gnome.org/show_bug.cgi?id=729426
+
+2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
+ Servers must handle Require headers and must report a failure
+ if they don't handle any of the Required options, see RFC 2326,
+ section 12.32: https://tools.ietf.org/html/rfc2326#page-54
+ https://bugzilla.gnome.org/show_bug.cgi?id=729426
+
+2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.3.1 ===
+
+2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.1
+
+2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From bcb1518 to 211fa5f
+
+2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ Update .gitignore
+
+2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/sessionmedia.c:
+ tests: fix memory leak in sessionmedia unit test
+
+2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: emit a signal before sending a message
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
+
+2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: pass context to send_message
+ Pass the current context to send_message, we will need it later.
+
+2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix typo in comment
+
+2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Do not stop thread twice if default_prepare() fails
+
+2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: set the watch to flushing before going to NULL
+ First set the watch to flushing so that we unblock any current and
+ future attempt to send data on the watch, Then set the pipeline to
+ NULL.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
+
+2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * tests/check/gst/sessionpool.c:
+ rtsp-session-pool: Fixes annotation
+ Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
+ in the sessionpool test.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
+
+2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: make media_prepare virtual
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
+
+2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: stop the thread in more error cases
+
+2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: allow NULL as the thread
+ Use the default context whan passing a NULL thread.
+
+2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: indent cleanup
+ Coverity was moaning about unreachable code, and I think it was just
+ confused by { being before the label. We'll see if it pops up again.
+ Coverity 1197705
+
+2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ client: Add drop-backlog property
+ When we have too many messages queued for a client (currently hardcoded
+ to 100) we overflow and drop the messages. Add a drop-backlog property
+ to control this behaviour. Setting this property to FALSE will retry
+ to send the messages to the client by waiting for more room in the
+ backlog.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
+
+2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: support for POST before GET when setting up a tunnel
+
+2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: remove watch of the second client after http tunnel setup
+ The second client will be freed after the HTTP tunnel has been set up.
+ Make sure it's RTSP watch is never dispatched again.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
+
+2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: Make media_prepare() fail if port allocation fails
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
+
+2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/media.c:
+ media test: cleanup the thread pool in tests
+
+2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: Unblock blocked streams in unprepare
+ The streams will be blocked when a live media is prepared.
+ The streams should be unblocked in gst_rtsp_media_unprepare.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
+
+2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: release the state lock when going to NULL
+ Set our state to UNPREPARING and release the state-lock before
+ setting the pipeline to the NULL state. This way, any pad-added
+ callback will be able to take the state-lock and check that we are now
+ unpreparing instead of deadlocking.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
+
+2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: protect status with lock
+ Make sure we only update the status with the lock.
+
+2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp: update for MIKEY API changes
+
+2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: parse the mikey response from the client
+ Parse the mikey response from the client and update the policy for
+ each SSRC.
+
+2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to set crypto info
+ Make a method to configure the crypto information of a stream.
+ Set udpsrc in READY instead of PAUSED so that we can configure caps
+ later.
+
+2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: cleanup error paths
+
+2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix docs
+
+2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-video.c:
+ test: enable SRTP only on RTSPS
+ We only want to enable SRTP when doing rtsp over TLS so that we can
+ exchange the keys in a secure way.
+
+2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-video.c:
+ test: print an error on failure
+
+2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * configure.ac:
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/Makefile.am:
+ stream: add SRTP support
+ Install srtp encoder and decoder elements in rtpbin
+ Add MIKEY in SDP
+
+2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/sessionpool.c:
+ tests: Add unit tests for sessionpool
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
+
+2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/threadpool.c:
+ tests: Improve code coverage of rtsp-threadpool tests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
+
+2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/sessionmedia.c:
+ tests: Improve code coverage for rtsp-session-media
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
+
+2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ gobject-introspection: Add annotations to support language bindings
+ In addition a few cosmetic changes:
+ * Adjust the order of arguments
+ * Fix typo: occured -> occurred
+ * Fix indentation after Return:-clauses
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
+
+2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't mix IPv4 and IPv6 addresses
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
+
+2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: take caps after the session manager
+ Take the caps for the SDP after they leave the rtpbin so that we can
+ also get the properties added by rtpbin elements.
+
+2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: release lock while pushing out packets
+ Keep a cache of the transports and use this to iterate the transport
+ while pushing packets. This allows us to release the lock early.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=725898
+
+2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: vmethod for modifying tunnel GET response
+ Add a vmethod tunnel_http_response where the response to the HTTP GET
+ for tunneled connections can be modified.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
+
+2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: make 1 media line per profile
+ If we have multiple profiles (AVP or AVPF) for a stream, make one m=
+ line in the SDP for each profile. The client is then supposed to pick
+ one of the profiles in the SETUP request. Because the m= lines have the
+ same pt, the client also knows that only 1 option is possible.
+
+2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ factory: add profile property and pass to media and streams
+
+2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-multicast.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: pass multicast connection for multicast-only stream
+ Pass the multicast address of the stream in the connection info in the
+ SDP so that clients try a multicast connection first.
+ Only allow multicast connections in the test-multicast example. Also
+ increase the TTL a little.
+
+2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * .gitignore:
+ .gitignore: Ignore gcov intermediate files
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
+
+2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: release some locks in error cases
+
+2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ docs: Enable and fix gtk-doc warnings
+ * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
+ * addresspool/mediafactory: Add missing annotation colon
+ * stream: Annotate return value
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
+
+2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From fe1672e to bcb1518
+
+2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 1a07da9 to fe1672e
+
+2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/Makefile.am:
+ examples: use LDADD for libs instead of LDFLAGS
+
+2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: make sure releases are in .doap file
+
+2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-cgroups.c:
+ examples: test-cgroups: don't put code with side effects into g_assert()
+ The g_assert() might get compiled out with the right
+ compiler/preprocessor flags.
+
+2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/.gitignore:
+ examples: add cgroup test binary to .gitignore
+
+2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-cgroups.c:
+ examples: fix cgroup test build
+ Fixes build failure caused by compiler warning:
+ test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
+
+2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ .gitignore: ignore temp files created in the course of 'make check'
+
+2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: don't loose frames handling new PLAY request
+ If client supplied a range check if the range specifies the start point.
+ If not, then do an accurate seek to the current position. If a start
+ point was specified do do a key unit seek to make sure the streaming
+ starts with decodeable frames.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
+
+2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Revert "media: only flush when setting a new start position"
+ This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
+ We need to do the flush in all cases, demuxer block currently for
+ non-flushing seeks.
+
+2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: only flush when setting a new start position
+ Only flush the pipeline when we change the start position with
+ a seek.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=724611
+
+2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set ttl-mc before adding the socket
+ Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
+ never be set on socket.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
+
+2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: stop thread if media is already prepared
+ in gst_rtsp_media_prepare() the thread is not used if media is already
+ prepared (e.g. media shared) so we want to stop the thread. otherwise, a
+ leak occurs.
+ https://bugzilla.gnome.org/show_bug.cgi?id=724182
+
+2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * Makefile.am:
+ build: Ship gst-rtsp-server.doap file
+
+2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Fix another compiler warning with gcc
+
+2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ rtsp-server: Fix lots of compiler warnings with clang
+
+2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * tests/Makefile.am:
+ configure: Synchronise with the configure scripts of the other modules
+
+2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
+
+2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ Revert "rtsp-server: support build against last stable release"
+ This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
+ Let us require 1.2.3 now, which is going to be released in a few
+ minutes.
+
+2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ session: improve RTP-Info
+ Ignore streams that can't generate RTP-Info instead of failing.
+ Don't return the empty string when all streams are unconfigured but
+ return NULL so that we don't generate and empty RTP-Info header.
+ Improve docs a little.
+
+2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ Don't free rtpinfo GString when it is NULL
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
+
+2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: only set keyframe flag when modifying start
+ Only set the keyframe flag when we modify the start position. The
+ keyframe flag should probably be ignored when no change is requested but
+ until we can claim this is all documented properly and all demuxer
+ implement this, avoid setting the flag.
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
+
+2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: Unref source after mainloop has quit to avoid races in GLib
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
+
+2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: handle NULL seqnum and rtptime arguments
+
+2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * tests/check/gst/threadpool.c:
+ thread-pool: Unref reused threads in gst_rtsp_thread_stop()
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
+
+2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: add fallback for missing stats property
+ Use a fallback when the payloader does not have a stats property
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
+
+2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From f7bc1c3 to 1a07da9
+
+2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: don't leak stats structure
+ Don't leak the stats structure and deal with NULL stats.
+
+2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Get rtpinfo properties atomically from payloader
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
+
+2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: refactor state change functions and signals
+ Make functions to set the target state and the pipeline state and emit
+ the signals from those functions.
+
+2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal to notify of pending state changes
+
+2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: support build against last stable release
+ Until 1.2.3 is out with the new get_type function and we
+ can require that.
+
+2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: fix compilation
+
+2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add property to configure profiles
+
+2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: let stream check supported transport
+ Delegate the check if a transport is allowed to the stream.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=720696
+
+2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to check supported transport
+ Add a method to check if a transport is supported
+
+2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure.ac: Only check for gstreamer-check, not check
+ We include check in gstreamer-check since quite some time now.
+
+2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: return clock-rate from get_rtpinfo
+ And use it to correct the rtptime to the requested start-time.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=712198
+
+2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ session-media: calculate start-time
+
+2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: also return the running-time
+ Return the running-time in the rtpinfo as well.
+
+2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ session-media: let the session-media make the RTPInfo
+ Add method to create the RTPInfo for a stream-transport.
+ Add method to create the RTPInfo for all stream-transports in a
+ session-media.
+ Use the session-media RTPInfo code in client. This allows us to refactor
+ another method to link the TCP callbacks.
+
+2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ mount-points: sort sequence before g_sequence_lookup
+ * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
+ sort sequence if dirty, otherwise lookup will fail.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
+
+2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: rename package from gst-rtsp to gst-rtsp-server
+ To match git module name and avoid confusion with the
+ rtsp lib in gst-plugins-base and rtsp plugin in -good.
+
+2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: bump core/base/good requirement to 1.2.0
+ Bump to released stable version and make implicit
+ requirements explicit.
+
+2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ * configure.ac:
+ Fix broken gettext setup which is not used anyway
+
+2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From dbedaa0 to d48bed3
+
+2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add setup_sdp vmethod
+ gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
+ gst_rtsp_media_setup_sdp.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
+
+2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Check return value of sscanf
+ streamid is only valid if sscanf matched something.
+
+2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix iteration
+ Wouldn't even enter the code block otherwise (i++ was used as the check
+ and not the postfix).
+
+2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add vmethod to configure media and streams
+ Implement a vmethod that can be used to configure the media and the
+ streams based on the current context. Handle the blocksize handling in
+ the default handler.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=720667
+
+2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ Make git ignore more unit test binaries
+
+2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * gst/rtsp-server/rtsp-token.h:
+ rtsp-server: add padding to many public structures
+ Not mini objects though, since they are not subclassable
+ anyway, nor kept on the stack or inlined in a structure.
+
+2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ media: add new create_rtpbin vmethod
+ * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
+ https://bugzilla.gnome.org/show_bug.cgi?id=719734
+
+2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
+
+ * tests/check/gst/media.c:
+ tests: fix memory leak, free test's thread pool
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
+
+2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ stream-transport: free url in finalize
+
+2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: also do state change in suspended state
+
+2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ media: also handle prepare and range in suspended state
+ When we are suspended, we are already prepared.
+ We can get the range in the suspended state.
+
+2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/sessionmedia.c:
+ check: add test for uri in setup
+ Added unit tests for the new functionality in GstRTSPStreamTransport.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
+
+2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: store setup uri and use in PLAY response
+ Store the uri used when doing the setup and use that in the PLAY
+ response.
+ fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
+
+2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ stream-transport: add method to get/set url
+
+2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: suspend after SDP and unsuspend before PLAYING
+ Based on patches by Ognyan Tonchev <ognyan@axis.com>
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * tests/check/gst/media.c:
+ * tests/check/gst/mediafactory.c:
+ media: add suspend modes
+ Add support for different suspend modes. The stream is suspended right after
+ producing the SDP and after PAUSE. Different suspend modes are available that
+ affect the state of the pipeline. NONE leaves the pipeline state unchanged and
+ is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
+ state and RESET will bring the pipeline to the NULL state.
+ A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
+ this means that the pipeline needs to be prerolled again.
+ Base on patches by Ognyan Tonchev <ognyan@axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: start live streams in blocked state
+ Start live streams in the blocked state and make them preroll using the
+ messages. This ensure that no data is played by the sink until we explicitly
+ unblock the stream right before going to PLAYING.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: refactor starting and waiting for preroll
+ Based on patches from Ognyan Tonchev <ognyan@axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add API to block streams
+ Add an API to block on the streams and make it post a message.
+ Based on patch by Ognyan Tonchev <ognyan@axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
+
+ * docs/libs/Makefile.am:
+ docs: Specify the override file
+ Even if it's empty (for now) it avoids make distcheck complaining
+
+2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: move default implementations to where they are used
+
+2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: take the right lock in gst_rtsp_media_set_pipeline_state()
+ We need to take the state_lock when calling this method.
+
+2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: handle add-added on non-bins too
+ Handle dynamic payloaders that are not bins, as used in the unit-test.
+
+2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media/-factory: Fix request pad name comments
+ These must be escaped for gtk-doc to parse the comments without warnings.
+
+2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ rtsp-media: remove transports if media is in error status
+ * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
+ trying to change to GST_STATE_NULL and media is in error status, we
+ remove all transports.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
+
+2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: use element metadata to find payloader
+ Use the element metadata to find the payloader instead of checking
+ for the base class.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
+
+2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ rtsp-stream: add getter for payload type
+ * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
+ * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
+ element and create the stream with this one instead of the dynpay%d
+ element.
+ https://bugzilla.gnome.org/show_bug.cgi?id=712396
+
+2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-token.c:
+ rtsp-*: Refer to NULL as a constant in comments
+ Plus one typo fix.
+ https://bugzilla.gnome.org/show_bug.cgi?id=714988
+
+2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ rtsp-*: Fix type name typos in comments
+ * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
+ * rtsp-auth: Refer to part of constant name as text
+ * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
+ * rtsp-session-media: Fix GstRTSPSessionMedia typo
+ * rtsp-stream: Fix typo when refering to GstBin
+ https://bugzilla.gnome.org/show_bug.cgi?id=714988
+
+2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * docs/README:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: Improve documentation
+ * Include annotation-glossary to quiet gtk-doc
+ * Rename remaining ClientState -> Context
+ * Rename object hierarchy file
+ * Remove stale chapter references
+ * Add missing function and object references
+ * Include missing GstRTSPAddressPoolResult
+ https://bugzilla.gnome.org/show_bug.cgi?id=714988
+
+2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: sprinkle some allow-none annotations for g-i
+
+2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to filter transports
+ Add a method to safely iterate and collect the stream transports
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
+
+2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp: allow NULL func in filters
+ Passing a null function make the filters return a list of
+ refcounted objects.
+
+2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * tests/check/gst/addresspool.c:
+ address-pool: fix address increment
+ Use a guint instead of guint8 to increment the address. It's still not
+ completely correct because a guint might not be able to hold the complete
+ address range, but that's an enhacement for later.
+ Add unit test to test improved behaviour.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708237
+
+2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ client: allow absolute path in requests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
+
+2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: make make_path_from_uri a vmethod
+
+2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/Makefile.am:
+ * tests/check/gst/stream.c:
+ stream: Add functions to get rtp and rtcp sockets
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
+
+2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-context.h:
+ context: defing a GType for the context
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
+
+2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ Fixed several GIR warnings
+
+2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: small typos
+
+2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/token.c:
+ tests: Add unit tests for token
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
+
+2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-token.c:
+ token: Validate args for gst_rtsp_token_is_allowed
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710520
+
+2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-token.c:
+ token: Fix bug when creating empty token
+ We always want to have a valid GstStructure in the token.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
+
+2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: avoid race in shutdown
+ If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
+ don't actually stop the mainloop ever. Solve this race by adding an idle source
+ to the mainloop that calls the _quit. This way we immediately exit the mainloop
+ if quit was called before we started it.
+
+2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/permissions.c:
+ tests: Add unit tests for permissions
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/mediafactory.c:
+ tests: Test mediafactory permissions
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: Fix refcounting when adding/removing roles
+ Previously a role that was removed was unreffed twice, and when
+ replacing an existing role the replaced role was freed while still being
+ referenced. Both bugs are now fixed.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/media.c:
+ * tests/check/gst/mediafactory.c:
+ * tests/check/gst/rtspserver.c:
+ tests: Check gst_rtsp_url_parse return value
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 865aa20 to dbedaa0
+
+2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: Fix socket leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=710088
+
+2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-session-pool: Make sure session IDs are properly URI-escaped
+ https://bugzilla.gnome.org/show_bug.cgi?id=643812
+
+2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * examples/.gitignore:
+ * examples/test-video.c:
+ examples: fix compilation when WITH_AUTH is defined
+ https://bugzilla.gnome.org/show_bug.cgi?id=710228
+
+2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * .gitignore:
+ gitignore: Add new test binary
+
+2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/threadpool.c:
+ thread-pool: Add unit test for the thread pools
+ https://bugzilla.gnome.org/show_bug.cgi?id=710228
+
+2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: Fix thread leak when reusing threads
+ https://bugzilla.gnome.org/show_bug.cgi?id=709730
+
+2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * tests/check/gst/rtspserver.c:
+ tests: fixed racy behavior in rtspserver tests
+ https://bugzilla.gnome.org/show_bug.cgi?id=710078
+
+2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/addresspool.c:
+ tests: Improve address pool unit tests
+ Add a range with mixed IPV4 and IPV6 addresses to pool.
+ Get an IPV4 address from an IPV6-only pool.
+ Get an IPV6 address from an IPV4-only pool.
+ Reserve a IPV6 address from an IPV4-only pool.
+ Check for unicast addresses in multicast-only pool.
+ Check for unicast addresses in uni-/multicast-mixed pool.
+ https://bugzilla.gnome.org/show_bug.cgi?id=710128
+
+2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: append query string in PAUSE/PLAY/TEARDOWN as well
+
+2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Add query to control path
+ If the SETUP url contains a query it must be appended to the control
+ path so that it matches any already created stream in the media. The
+ query will also be appended to the session media path.
+
+2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: remove old line
+
+2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Correct control comparison
+ https://bugzilla.gnome.org/show_bug.cgi?id=709176
+
+2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Check dynamically if the pipeline supports seeking
+ We should not depend on whether or not the pipeline state change
+ returned NO_PREROLL or not. A media could dynamically change its
+ element and switch from seekable to non seekable so it's best to test
+ the seekable nature of the pipeline dynamically when we try to do a seek.
+
+2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Return FALSE if seeking is not supported
+
+2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: don't seek accurate by default
+ Accurate seeking is perhaps a little overkill in the most common situation and
+ causes some formats (mp3) over slow media to seek extremely slowly.
+
+2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: fix unit test
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
+
+2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Reply 400 if media cannot be constructed
+ Reply 400 Bad Request instead of 503 Service Unavailable if media
+ cannot be constructed in SETUP.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
+
+2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Send setup reply once only
+ If find_media() failed in handle_setup_request() two replies was sent.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
+
+2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6b03ba7 to 865aa20
+
+2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: Emit client-connected signal earlier
+ Emit client-connected before the client ref is given to a GSource,
+ otherwise client-connected can be emitted after the client object has
+ been freed.
+
+2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/addresspool.c:
+ addresspool: return reason of failure
+ Let gst_rtsp_address_pool_reserve_address() return the reason why
+ the address could not be reserved.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
+
+2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
+
+ * autogen.sh:
+ autogen.sh: Sync behaviour with other GStreamer modules
+ Allows building from outside of tree amongst other things
+
+2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
+
+ * common:
+ Automatic update of common submodule
+ From b613661 to 6b03ba7
+
+2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 74a6857 to b613661
+
+2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 01a7a46 to 74a6857
+
+2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Do not read beyond end of path string
+ If the setup was done without a control url, make sure we don't try to read the
+ non-existing control string and crash.
+
+2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Fix RTPInfo header
+ Refactor the method to make the content_base.
+ Use the content-base and the control url to construct the RTPInfo
+ url.
+
+2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: map url to path only in describe
+ Only map the request url to a path in the DESCRIBE method. The SDP then
+ contains the base and control urls that should be used to SETUP/PAUSE/
+ PLAY/TEARDOWN the media.
+
+2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Revert "client: map URL to path in requests"
+ This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
+ This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
+ contains the base and control urls which are used in the SETUP, PLAY,
+ PAUSE and TEARDOWN requests.
+
+2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: map URL to path in requests
+
+2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ mount-points: make vmethod to make path from uri
+ Make a vmethod to transform an url into a path. The path is then used to lookup
+ the factory. This makes it possible to also use other bits of the url, such as
+ the query parameters, to locate the factory.
+
+2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: Add cleanup to wait for the threadpool to finish
+ Also fix race condition if two threads are asking for the first
+ thread from the thread pool at once. This would case two internal
+ GThreadPools to be created.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707753
+
+2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ client: free threadpool
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * tests/check/gst/mountpoints.c:
+ mountpoints tests: unref matched factories
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * tests/check/gst/media.c:
+ media tests: unref thread pool and caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ auth, media, media-factory: unref permissions
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ Makefile: add rule for appsrc example
+
+2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-appsrc.c:
+ tests: add appsrc example
+ Add an example on how to use appsrc to feed the server pipeline with data.
+
+2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: remove query part from content-base string
+ Make sure that after the control url has been resolved, it's
+ not a part of the query-string.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
+
+2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: don't check url in response
+ There is no url or method in the response to check
+
+2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Add handle-response signal for when we receive a GET_PARAMETER response
+
+2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Fix gst_rtsp_server_client_filter, using wrong variable type
+
+2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
+ For AAC we need to check for framed=true instead of parsed=true.
+ https://bugzilla.gnome.org/show_bug.cgi?id=701384
+
+2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: optimize pipeline for protocols
+ When TCP is not an allowed protocol for the stream, avoid creating the
+ appsrc/appsink/queue and tee elements.
+
+2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: set protocols on streams
+
+2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use protocols supported by stream
+
+2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ media-factory: allow all protocols
+
+2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: configure protocols in new streams
+
+2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add protocols property
+
+2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: send state in "new-state" signal
+ https://bugzilla.gnome.org/show_bug.cgi?id=705110
+
+2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
+
+ * configure.ac:
+ build: add subdir-objects to AM_INIT_AUTOMAKE
+ Fixes warnings with automake 1.14
+ https://bugzilla.gnome.org/show_bug.cgi?id=705350
+
+2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add method to iterate clients of server
+
+2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add vmethod for rtsp-media subclass to access rtpbin
+
+2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ small documentation fix
+
+2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Do not take range header if range is invalid
+
+2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media.c:
+ media: add docs for new method
+
+2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add API to rtsp-media set the pipeline's state
+
+2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Update current position/duration when gst_rtsp_media_get_range_string is called
+
+2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-cgroups.c:
+ tests: add some more docs
+
+2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * tests/check/gst/client.c:
+ ClientState -> Context
+ Rename the clientstate to context and put the code in a separate file.
+
+2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ auth: add support for default token
+ The default token is used when the user is not authenticated and can be used to
+ give minimal permissions.
+
+2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: use defines when possible
+
+2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ address-pool: improve docs
+
+2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: add the role to the copy
+
+2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: Also copy the roles
+
+2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: Make it build
+
+2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.h:
+ docs: small fixes
+
+2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ docs: improve docs
+
+2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * tests/check/gst/addresspool.c:
+ * tests/check/gst/rtspserver.c:
+ address-pool: cleanups
+ Remove redundant method, improve docs.
+
+2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-token.c:
+ docs: improve docs
+
+2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: implement _remove_role
+
+2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: update docs
+
+2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: simplify tests
+ Client settings are now disabled by default so we don't need an auth
+ module to disable them.
+
+2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: add default authorizations
+ When no auth module is specified, use our table of defaults to look up the
+ default value of the check instead of always allowing everything. This was
+ we can disallow client settings by default.
+
+2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ README: update readme
+
+2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: add more docs
+
+2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: fix race in thread reuse
+ If we try to reuse a thread right after we made it stop, we end up using a
+ stopped thread. Catch this case and only reuse threads that are not stopping.
+
+2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: add small debug
+
+2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ client: fix test
+ Add some permissions to media so we can use the auth and enable
+ client settings.
+
+2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: support pushed context in handle_request
+ If we already have a pushed state, reuse it and add our own things. This makes
+ it easier to write tests.
+
+2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: don't auth on methods
+ Don't authorize on methods anymore but on the resources that we
+ try to access, this is more flexible.
+ Move the authorization checks to where they are needed and let the
+ check return the response on error.
+
+2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mount-points: add some debug
+
+2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: almost fix test
+
+2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ auth: let the auth module check client_settings
+ Let the auth module decide if client settings are allowed for the
+ current client.
+
+2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ token: add method to check boolean permission
+
+2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ token: simplify token constructor
+ Use variable arguments to make easier API.
+
+2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add convenience API for factory
+
+2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ permissions: simplify API a little
+ Avoid passing GstStructure in the add_role method, use varargs instead
+ to construct the structure behind the scenes. We can then also use the
+ structure name as the role and simplify some more logic.
+
+2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: fix typo
+
+2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ auth: handle unauthorized response
+ Move handling of the unauthorized response to the auth module, it can add
+ the appropriate headers to request authorization for the required method
+ much better than the client.
+
+2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: allow for sending any message, not only requests
+ Change the _send_request() method to _send_message() so that we
+ can both send requests and replies.
+
+2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-server.h:
+ docs: fix docs
+
+2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ auth: move TLS handling to auth module
+ Remove the TLS settings on the server and move it to the auth module because
+ that is where security related bits go.
+
+2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add state push/pop
+
+2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add connection to state
+
+2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mount-points: fix debug
+
+2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/media.c:
+ tests: fix media test
+
+2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: we don't require a state
+
+2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: let context ref the server
+ So that we don't risk losing the server object early anc crash.
+
+2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: fix client test
+
+2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-token.c:
+ docs: improve docs
+
+2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ session-pool: make vmethod to create a session
+ Make a vmethod to create a sessions so that subclasses can create
+ custom session objects
+
+2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ docs: more updates
+
+2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ docs: update docs
+
+2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * examples/Makefile.am:
+ configure: compile cgroup example conditionally
+ Only compile the cgroup example when we have libcgroup
+
+2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * examples/Makefile.am:
+ * examples/test-cgroups.c:
+ examples: add cgroups example
+
+2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/rtspserver.c:
+ tests: fix compilation
+
+2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: fix vmethod invocation
+
+2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: store thread type in thread
+
+2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: pass thread from pool to media _prepare
+ Get a thread from the configured threadpool and pass it to the prepare method of
+ the media.
+
+2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: Accept a thread in _prepare
+ Remove out own threadpool handling and use the provided thread and
+ maincontext for the bus messages and the state changes.
+
+2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: configure client thread pool
+
+2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add method to configure thread pool
+
+2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: use thread pool
+ Use the thread pool instead of doing our own thing.
+
+2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: add object to manage threads
+ Add an object to manage the client and media threads.
+
+2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: debug authorization check
+
+2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: start media pipeline in context
+ Start the media pipeline in the provided context (or our default one
+ when NULL). This makes sure that we run the bus thread in this context and that
+ all media threads are children of this context.
+
+2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: pass permissions to media by default
+
+2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ test: add permissions to auth test
+ Ass some permissions to the media factory in the test.
+
+2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ auth: simplify auth checks
+ Remove client from methods, it's now in the state
+ Perform the check specified by the string, use the information from the
+ thread local context.
+
+2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add state to current thread
+ Add the client to the ClientState object.
+ Place the ClientState on the current thread.
+
+2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: make it possible to set permissions
+ Make it possible to set permissions on media and media factory objects
+
+2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ permissions: add permissions object
+ Add a mini object to store permissions based on a role.
+
+2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ auth: add auth checks
+ Add an enum with auth checks and implement the checks in the auth object.
+ Perform the checks from the client.
+
+2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ auth: use the token after authentication
+ After we authenticated a user, keep the Token around in the state.
+
+2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * tests/check/gst/media.c:
+ media: add optional context for bus messages
+ Add an optional mainloop to _prepare that will handle the bus messages instead
+ of always using the shared mainloop.
+
+2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ token: add authorization token
+ Add a simply miniobject that contains the authorizations. The object contains a
+ GstStructure that hold all authorization fields. When a user is authenticated,
+ the auth module will create a Token for the user. The token is then used to
+ check what operations the user is allowed to do and various other configuration
+ values.
+
+2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ auth: remove auth from media and factory
+ Remove the auth object from media and factory. We want to have the RTSPClient
+ authenticate and authorize resources, there is no need to place another auth
+ manager on the media/factory.
+
+2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ auth: add support for multiple basic auth tokens
+ Make it possible to add multiple basic authorisation tokens to one authorization
+ object. Associate with each token an authorization group that will define what
+ capabilities are allowed.
+
+2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: error out on non-aggregate control
+ We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
+
+2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: rework setup request a little
+ Cache the media in DESCRIBE based on the longest matching path with the uri
+ that we can find in the mount points.
+ Rework the setup request a little to get the media from the session or from
+ the longest matching path, this way we can derive the control string as
+ everything after the path instead of hardcoding it.
+ Find the stream based on the control string and only open a session when all
+ this can be done.
+
+2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add method to find a stream by control url
+
+2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to check control url of stream
+
+2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: use path matching for session media
+ Use a path string instead of a uri to lookup session media in the sessions. Also
+ use path matching to find the largest possible path that matches.
+
+2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * tests/check/gst/mountpoints.c:
+ mount-points: remove useless vmethod
+ Making lookups in the mount points should not be done with a URL, if there is a
+ mapping to be done from URL to mount points, we'll need to do it somewhere
+ else.
+
+2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * tests/check/gst/mountpoints.c:
+ mount-points: improve mount point searching
+ Use a GSequence to keep track of the mount points.
+ Match a URL to the longest matching registered mount point. This should be the
+ URL to perform aggreagate control and the remainder is the stream specific
+ control part.
+ Add some unit tests for this.
+
+2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Allow building of static library
+
+2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/mediafactory.c:
+ tests: fix compilation
+
+2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: get control string from stream
+ Use the control string as configured in the stream.
+
+2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add methods and property to set control string
+
+2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: cleanups
+ Rename variables for clarity
+ Keep media in state when we can
+
+2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add more support for IPv6
+ Rename _get_address to _get_multicast_address in GstRTSPStream to
+ make it clear that this function only deals with multicast.
+ Make it possible to have both an IPv4 and IPv6 multicast address on
+ a stream. Give the client an IPv4 or IPv6 address depending on the
+ address it used to connect to the server.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
+
+2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix comment
+
+2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: handle failed port allocation
+ Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
+ can't allocate any family at all. Also keep track of what port families we
+ allocated.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
+
+2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: improve docs
+
+2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ stream-transport: remove old if 0 block
+
+2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/client.c:
+ tests: fix tests
+ gst_rtsp_client_get_uri() has been removed
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
+
+2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add method to filter managed sessions
+ Add a method to filter the sessions managed by this client connection.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=703016
+
+2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: remove _get_uri() method
+ Remove the get_uri() method on the client. A client has no uri, the uri
+ property is an internal property to manage the last cached media for
+ the client.
+
+2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: fix typo
+
+2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Do not leak the query in default_query_stop
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
+
+2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't unlock when conversion fails
+ Don't unlock the state lock when conversion fails because it was not locked.
+
+2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add query_position and query_stop vmethods to rtsp-media
+
+2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Fix typo in property install for rtsp-media's time-provider
+
+2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: clean some variables
+ Clean some variables and add some guards to _send_request()
+
+2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Add gst_rtsp_client_send_request API
+ This makes it possible to send arbitrary messages to a client, such as
+ SET_PARAMETER or GET_PARAMETER
+
+2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add _get_element() method
+ Add method to get the element used when creating the media.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
+
+2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix docs
+
+2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: allow access to the rtp session
+ https://bugzilla.gnome.org/show_bug.cgi?id=703004
+
+2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ dscp qos support in gst-rtsp-stream
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
+
+2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/rtspserver.c:
+ tests: fix test
+ Actually do what the comment says. Also keep the old code around, not sure what
+ should happen when you get a 454 from a TEARDOWN, does it close the connection?
+ it currently doesn't.
+
+2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: also watch newly created session
+ When we newly created a session, start watching it immediately instead of
+ on the next request.
+
+2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/client.c:
+ tests: add unit test for new-session
+ See https://bugzilla.gnome.org/show_bug.cgi?id=701587
+
+2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: emit new-session when new session is created
+ Only emit new-session when we created a new session for a client, not when a
+ client picked up a previous session.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
+
+2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: handle asterisk as path in requests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
+
+2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: handle segment query format mismatch
+ It's possible that the segment query returns with a different format than what
+ we asked for, handle this case also.
+
+2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: use segment stop in collect_media_stats
+ Use segment stop instead of duration as range end point.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
+
+2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: Do not leak the element in take_pipeline
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
+
+2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: Make configure_client_transport virtual
+ This patch makes configure_client_transport virtual. The functionality is
+ needed to handle some weird clients sending multicast transport settings as url
+ options.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
+
+2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: Make param_set and param_get virtual
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
+
+2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: convert_range replaces get_range_times
+ get_range_times worked for handling UTC ranges for seeks, but we also
+ need to convert back from NPT to the requested unit in
+ get_range_string. convert_range is now used for both.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
+
+2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: cleanup sdp info
+ We don't need to pass the proto, we can more easily check a boolean.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
+
+2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ use 0.0.0.0 or :: for c= line instead of server address
+
+2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ use local address, not remote, in SDP
+ See https://bugzilla.gnome.org/show_bug.cgi?id=702063
+
+2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 098c0d7 to 01a7a46
+
+2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: possibility to override range time conversion
+ Make it possible to override the conversion from GstRTSPTimeRange to
+ GstClockTimes, that is done before seeking on the media
+ pipeline. Overriding can be useful for UTC ranges, where the default
+ conversion gives nanoseconds since 1900.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
+
+2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: Expose the use_client_settings API
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
+
+2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtspstream: handle both ipv4 and ipv6 clients
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
+
+2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
+ This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
+ We already have a way to place extra attributes in the SDP by using a string
+ property with prefix x- or a- in the caps.
+
+2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
+ This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
+ We already have a way to place extra attributes in the SDP, just make a string
+ property in the payloader with a- or x- prefix.
+
+2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp: place a- and x- properties as attributes
+ application/x-rtp has properties with a- and x- prefixes that should be
+ placed as attributes in the SDP for the media instead of being added to the
+ fmtp.
+
+2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-video.c:
+ example: add TLS example
+
+2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add support for TLS
+ Add methods to set and get a TLS certificate.
+ Add vmethod to configure a new connection. By default, configure the TLS
+ certificate in a new connection if needed.
+
+2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: remove accept_client vmethod
+ This vmethod is not very useful so remove it.
+
+2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: don't crash on NULL GError
+
+2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-session-pool: corrected session timeout detection
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
+
+2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve debug
+
+2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ server: refactor connection setup
+ Let the server accept the socket connection and construct a GstRTSPConnection
+ from it. Remove the code from the client and let the client only deal with
+ a fully configure GstRTSPConnection object.
+ We will need this later when the server will configure the connection for
+ TLS.
+
+2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: keep the transport object alive
+ Keep the transport object alive while we have it as qdata on the
+ source.
+
+2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: Do not crash on nmapping of server
+ * generate error when gst_rtsp_connection_accept fails
+ * do not stop accepting incoming connections because
+ accepting a client fails
+ https://bugzilla.gnome.org/show_bug.cgi?id=701072
+
+2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
+ https://bugzilla.gnome.org/show_bug.cgi?id=700953
+
+2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Parse framerate caps field and set SDP attribute
+ The SDP attribute and its format is described in RFC4566.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
+
+2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Parse width/height from caps and set SDP attribute
+ The SDP attribute and its format is described in RFC6064.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
+
+2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ * tests/check/gst/client.c:
+ rtsp-sdp: add bandwidth line
+ https://bugzilla.gnome.org/show_bug.cgi?id=699220
+
+2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 5edcd85 to 098c0d7
+
+2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/media.c:
+ tests: add dynamic payloader prepare/unprepare check
+
+2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: release lock when removing fakesink
+
+2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set elements to NULL before removing
+ When removing a stream, set the elements to NULL first. This avoids
+ element-is-not-in-NULL-state errors when we dispose the elements.
+
+2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 3cb3d3c to 5edcd85
+
+2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: listen to pad-removed signals
+ Listen to the pad-removed signal and remove the stream associated with the
+ removed pad.
+ Add signal to be notified of the removed pad.
+ Remove the fakesink in unprepare()
+ Fix signatures of the signal methods
+
+2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-sdp.c:
+ tests: add example of reusable pipelines
+
+2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to get the srcpad
+
+2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/media.c:
+ check: add media prepare/unprepare test
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: disconnect from signal handlers in unprepare()
+ We connected to the pad-added and no-more-pads signals in prepare() so
+ we need to disconnect from them in unprepare().
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't free streams array
+ Don't free the streams array in the unprepare() method, they were not
+ added in prepare().
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't unref the pipeline in unprepare
+ Unprepare() should undo what prepare() does. Because the pipeline is
+ not created in prepare(), we should not unref it in unprepare()
+
+2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: clear session and caps for reuse
+ Set the session and caps to NULL after unref otherwise we might unref
+ them again later.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: send out teardown signal before tearing down
+ The advantage is that in the signal handler you get direct access to
+ information about what streams are about to get torn down (in the
+ GstRTSPClientState).
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
+
+2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: expose connection
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
+
+2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From aed87ae to 3cb3d3c
+
+2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ media: add method to get the base_time of the pipeline
+ Together with a shared clock, this base-time could eventually be sent to
+ the client so that it can reconstruct the exact running-time of the clock
+ on the server.
+
+2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ media: add GstNetTimeProvider support
+ Add a property to let the media provide a GstNetTimeProvider for its clock.
+ Make methods to get the clock and nettimeprovider
+ Add a x-gst-clock property to the SDP with the IP and port number of the nettime
+ provider and also the current time of the clock. This should make it possible
+ for (GStreamer) clients to slave their clock to the server clock.
+
+2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 04c7a1e to aed87ae
+
+2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: wait for buffering to complete
+ Wait for buffering to complete before changing the state to the target state.
+
+2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: small cleanup
+
+2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: remove extra unref in test_setup_non_existing_stream
+ The unref is not needed anymore, teardown runs without it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=696542
+
+2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: GSocketService cleanup in test_bind_already_in_use
+ Use g_socket_service_stop so the rtspserver test stops listening for
+ incoming connections in test_bind_already_in_use.
+ https://bugzilla.gnome.org/show_bug.cgi?id=696541
+
+2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
+ Instead use a GWeakRef which is safe to use
+ This is a known GLib bug, see:
+ https://bugzilla.gnome.org/show_bug.cgi?id=667145
+
+2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * tests/check/gst/media.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-media/client: Reply to PLAY request with same type of Range
+ Remember the type of Range from the PLAY request and use the same type for
+ the reply.
+
+2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * tests/check/gst/client.c:
+ rtsp-client: expose uri
+
+2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/mediafactory.c:
+ tests: Hold ref while creating second media
+ To test if the media aren't shared, make sure we keep the first one while creating a second
+ otherwise the same memory address may be reused.
+
+2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * configure.ac:
+ configure: remove out-of-date comment
+
+2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * .gitignore:
+ .gitignore: ignore more build files
+
+2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * tests/check/Makefile.am:
+ tests: use right _LIBS variable for gst-plugins-base libs
+
+2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ check: add librtp to libs
+
+2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Add test to check selecting a port the server will send from
+
+2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Make sure packets are actually received
+
+2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Select unicast address from pool if appropriate
+
+2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Properties are always there in Gst 1.0
+
+2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/addresspool.c:
+ tests: Add tests for unicast addresses in pool
+
+2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * tests/check/gst/addresspool.c:
+ address-pool: Verify that multicast addresses are used for multicast and vice-versa
+
+2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/addresspool.c:
+ address-pool: Add unicast addresses
+
+2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * configure.ac:
+ * gst/rtsp-server/rtsp-server.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-server: Limit the number of threads per server instance
+ If we exceed the maximum, just round robin the clients over the existing
+ threads.
+
+2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: No need to store the GMainContext in the client context
+
+2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Add test for client disconnection
+
+2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test client and session timeouts with multiple threads
+
+2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ Document locking and its order
+
+2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test that slow DESCRIBE don't block other clients
+
+2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/client.c:
+ tests: Add tests for client-requested multicast address
+
+2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: Put the various functions in the right sections
+
+2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ docs: Generate docs for GstRTSPAddressPool
+
+2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ client: Check client provided addresses against the address pool
+
+2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * tests/check/gst/addresspool.c:
+ address-pool: Add API to request a specific address from the pool
+ Also add relevant unit tests.
+
+2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/mediafactory.c:
+ tests: Check the passing around of a RTSPAddressPool
+ Make sure the RTSPAddressPool is propagated from the MediaFactory all the
+ way down to the stream.
+
+2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/addresspool.c:
+ tests: Add more tests for the address pool
+
+2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ address-pool: Fix off by one error
+ When splitting a port range, the port after a skip is not part of range.
+
+2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 2de221c to 04c7a1e
+
+2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
+
+ * configure.ac:
+ configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
+ AM_CONFIG_HEADER was removed in automake 1.13
+ https://bugzilla.gnome.org/show_bug.cgi?id=693368
+
+2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From a942293 to 2de221c
+
+2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: make sure the watch exists while sending data
+ Protect the send_func with a lock. This allows us to wait for sending
+ to complete before changing the send_func and user_data. We add an
+ extra ref to the watch to make sure that it remains valid during
+ sending.
+ When closing the connection, set the send_func to NULL
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
+
+2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ tests: use GST_*_1_0 environment variables everywhere
+ The _1_0 suffixed environment variables override the
+ non-suffixed ones, so if we're in an environment that
+ sets the _1_0 suffixed ones, such as jhbuild, we need
+ to set those to make sure ours actually always get
+ used.
+
+2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From acb04d9 to a942293
+
+2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: set the client backlog
+ Set the client backlog to a reasonable default
+
+2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Make the element a constructor parameter
+ https://bugzilla.gnome.org/show_bug.cgi?id=689594
+
+2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * docs/libs/Makefile.am:
+ docs: Link with gcov library when gcov is enabled
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
+
+2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: match prepare with unprepare
+ Really unprepare when there were an equal amount of prepare calls.
+
+2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: media has to be unprepared in finalize
+ Because unprepare takes away the last ref on the media.
+
+2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
+ This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
+ We can't use the refcount to trigger unprepare because it is the unprepare call
+ that removes the last refcount after all messages are consumed. What we should
+ probably do is make a prepared refcount and only unprepare when the refcount
+ reaches 0.
+
+2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: let the source unref the last media ref
+ the last ref to the media is held by the source so we don't need to add more ref
+ and unrefs, we simply destroy the media when the source is gone.
+
+2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: improve debug
+
+2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: check state
+ Make sure we are in the right state when collecting the position and duration.
+ Only make ourselves PREPARED when we were previously PREPARING.
+
+2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: use g_object_ref/unref for GObjects
+
+2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
+ Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
+ GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
+ isn't being used anymore.
+
+2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Fix compiler warning
+
+2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
+
+2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-media.h:
+ small cleanup
+
+2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: avoid element leak
+
+2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: require an element in media constructor
+
+2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Revert "client: TEARDOWN brings that state to Init again"
+ This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
+ The object is already disposed, there is no point in setting the state.
+
+2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: TEARDOWN brings that state to Init again
+
+2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/media.c:
+ rtsp: make object details private
+ Make all object details private
+ Add methods to access private bits
+
+2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/media.c:
+ tests: add media tests
+
+2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: check if prepared for some methods
+ Check that the media object is prepared before doing seek and getting the
+ current position etc.
+ Add some g_return checks.
+
+2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/mediafactory.c:
+ tests: add mediafactory test
+
+2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: improve debug
+
+2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: unref pipeline in finalize to avoid leaking it
+
+2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp: use gst_object_unref on GstObjects
+
+2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: require an url
+
+2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ examples: fix include
+
+2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.h:
+ server: remove unused include
+
+2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/mountpoints.c:
+ tests: add test for mountpoints
+
+2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix factory leak
+ Keep the factory in the state object only for authorization checks and make
+ sure we unref it on failure. Also don't keep invalid objects in the state
+ object.
+
+2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mounts: add g_return_if guards
+
+2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: add more tests
+
+2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve debug
+
+2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve debug and fix leaks
+ Cleanup the uri and session when there is a bad request.
+
+2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * common:
+ update common
+
+2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ test: add test for session in options request
+
+2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use 454 when session can't be found
+ We should use 454 when a session can't be found because there was no session
+ pool configured in the server. This is not a server configuration problem
+ because the server on which the request is done might not be the same one that
+ will keep the sessions for us and so it does not need to support sessions.
+
+2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: only free connection when there is one
+ It's possible that the client doesn't have a connection when we try to free it.
+
+2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/client.c:
+ tests: add unit test for the client object
+
+2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: small cleanup
+
+2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ client: remove unused include
+
+2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix compilation
+
+2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: call destroy without the lock
+
+2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: make the client usable without a socket
+ Make a method to let the client handle a message and a callback when the client
+ wants us to send a response message back. This makes it possible to also use the
+ client object without the sockets, which should make it easier to test.
+
+2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: small cleanup
+
+2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ client: remove reference to server
+ We don't need to keep a ref to the server
+
+2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add locking
+ Also add some g_return_if()
+
+2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: log more errors
+
+2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix compilation
+
+2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add generic close-after-send support
+ Add a property to send_response() to close the connection after the response has
+ been sent to the client.
+
+2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * docs/libs/gst-rtsp-server.types:
+ * examples/test-auth.c:
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-multicast.c:
+ * examples/test-multicast2.c:
+ * examples/test-ogg.c:
+ * examples/test-readme.c:
+ * examples/test-sdp.c:
+ * examples/test-uri.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * tests/check/gst/rtspserver.c:
+ MediaMapping -> MountPoints
+ Describes better what the object manages.
+
+2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump required version of -base
+
+2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix seeking
+
+2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: support more Range formats
+ Use the new -base methods to convert the Range string into a seek start and stop
+ value.
+
+2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-launch.c:
+ examples: fix whitespace
+
+2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ test-auth: add example of how to remove sessions
+ Add an example of the session filter api.
+
+2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ test-uri: remove mapping example
+
+2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ test-uri: fix callback signature
+
+2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: keep ref to factory while media active
+ While the media from a factory is alive, keep a ref to the factory.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
+
+2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: add some debug
+
+2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set udp sources to PLAYING
+ Set the UDP sources to PLAYING and locked state before we add it to the pipeline
+ so that it doesn't cause our pipeline to produce ASYNC-DONE.
+
+2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: take ref to factory
+ Take a ref to the factory that we place in our list.
+
+2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/Makefile.am:
+ * tests/test-reuse.c:
+ test: add test for server reuse
+ See https://bugzilla.gnome.org/show_bug.cgi?id=688395
+
+2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: start and stop multiple times
+ Stop listening on the RTSP port when the GSource is removed, so clients
+ can't connect and the server can be started again.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
+
+2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: fix small leak
+
+2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: unref source in finish_unprepare
+ The source is created in prepare, unref it in finish_unprepare.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=688707
+
+2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: remove bus watch before finalizing
+ * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
+ * An extra media ref is added for the bus watch. This extra ref is unreffed by
+ the GDestroyNotify function.
+ * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
+ * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
+ gst_rtsp_media_unprepare before unreffing the media.
+ This way, the bus watch will be removed before the media is finalized.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
+
+2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: wait until the TEARDOWN response is sent to close the connection
+ Responses can be sent async so we need to wait until the TEARDOWN response has
+ been written before we close the connection to the client. This avoids the risk
+ of writing/polling closed sockets.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
+
+2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: plug socket leak
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
+
+2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6bb6951 to a72faea
+
+2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-server: don't use deprecated API
+
+2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix unused-but-set-variable compiler warning
+ rtsp-client.c:1260:21: error: variable 'protocols' set but not used
+
+2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * TODO:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp: cleanups
+
+2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-multicast2.c:
+ examples: add another multicast example
+ Add an example for how to configure separate multicast ranges for each media
+ stream.
+
+2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-multicast.c:
+ test: set shared
+
+2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ stream: use the address managed by the stream
+ Use the address managed by the stream for multicast. This allows us to have 1
+ multicast address for each stream.
+ Because the address is now managed by the stream we don't have to pass it around
+ anymore.
+ Set the address pool on the streams.
+
+2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp: improve debug
+
+2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal for new streams
+ This allows applications to listen for new streams and configure properties on
+ them, like the address pool.
+
+2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: configure address pool in new streams
+
+2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add methods to deal with address pool
+ Add methods to get and set the address pool for the stream
+ Add method to allocate and get the multicast addresses for this stream.
+
+2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: remove MTU property
+ It is a stream property
+
+2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: set blocksize only on stream
+ Set the blocksize only on the current stream.
+
+2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: share src and sink sockets
+ the allocated socket is in the used-socket property, not socket.
+
+2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * tests/check/gst/addresspool.c:
+ rtsp: make address-pool return an address object
+ Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
+ store more info in the structure and allows us to more easily return the address
+ to the right pool when no longer needed.
+ Pass the address to the StreamTransport so that we can return it to the pool
+ when the stream transport is freed or changed.
+
+2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-multicast.c:
+ examples: add multicast example
+ Show how to set up the multicast address pool so that media can be
+ server with multicast.
+
+2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp: use AddressPool
+ Remove the multicast_group property.
+ Use the configured addresspool to allocate multicast addresses.
+
+2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ address-pool: add clear method
+
+2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ address-pool: small cleanups
+
+2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/addresspool.c:
+ tests: add addresspool unit test
+
+2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ address-pool: add object to manage multicast addresses
+ Make an object that can manage a rage of multicast addresses and ports.
+
+2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: set default max-threads property
+
+2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: wait for concurrent _prepare
+ If a prepare is busy, wait for the result.
+
+2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: add lock around message handler
+ We don't want to dispatch messages while we are still processing the result of
+ the state change.
+
+2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add lock to protect state changes
+
+2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add locking
+
+2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream-transport: add keep-alive method
+
+2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream-transport: add method to handle RTP/RTCP
+ Call new methods instead of poking into the structures directly.
+
+2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ session-media: add locking
+
+2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: add locking
+
+2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: free old socket
+
+2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ mapping: add locking
+
+2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: add locking
+
+2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ auth: add locking
+
+2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add max-thread property
+
+2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: use a threadpool for the mainloops
+
+2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: rename method
+ gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
+ don't really create the client from the socket, we use the socket for the
+ client.
+
+2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ server: rework maincontext handling in clients
+ Make a separate method to attach a client to a MainContext.
+ Let the server decide in what GMainContext the client will operate and give this
+ context to the client in attach. Then the server can later decide to use a
+ separate thread for each client or just use the mainthread.
+
+2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: move session header code in session object
+
+2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * COPYING:
+ * COPYING.LIB:
+ * examples/test-auth.c:
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-readme.c:
+ * examples/test-sdp.c:
+ * examples/test-uri.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/rtspserver.c:
+ * tests/test-cleanup.c:
+ Fix FSF address
+
+2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-server: added annotations to indicate type of ownership transfer of return values
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * configure.ac:
+ No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
+
+2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * Makefile.am:
+ * bindings/Makefile.am:
+ * bindings/vala/Makefile.am:
+ * bindings/vala/gst-rtsp-server-0.10.deps:
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.deps:
+ * bindings/vala/packages/gst-rtsp-server-0.10.files:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
+ * configure.ac:
+ bindings: remove vala bindings
+ They'll be reunited with the other GStreamer bindings
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ rtsp: only create transport when needed
+ Only create the StreamTransport when configured.
+
+2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: small cleanup
+
+2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ rtsp: refactor configuration of transport
+ Move the configuration of the transport to a place where it makes
+ more sense.
+
+2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: refactor transport parsing
+
+2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: refuse to change the MTU on shared media
+ If we change the MTU of chared media, it changes for all clients.
+ We don't want to set the MTU to something large for clients that
+ stream over UDP.
+
+2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-mp4.c:
+ * gst/rtsp-server/rtsp-media.c:
+ small fixes to docs and debug
+
+2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: transports must already have been removed
+
+2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: improve join and leave of the pipeline
+ simplify code
+ Do the cleanup properly
+ Add some docs
+
+2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: move unprepare below default implementation
+ Makes it easier to find the default implementation
+
+2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: signal unprepared when we actually finish
+
+2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: no need to unlock, unprepare does that when needed
+
+2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ docs: update docs
+
+2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp: fix MTU setting
+ Fix setting of the MTU. There is no need for a vmethod.
+
+2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ docs: update docs
+
+2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump version number after refactoring
+
+2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp: massive refactoring
+ Make GObjects from the remaining simple structures.
+ Remove GstRTSPSessionStream, it's not needed.
+ Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
+ Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
+ a GstRTSPStream should be transported to a client.
+ Rename GstRTSPMediaFactory::get_element -> create_element because that
+ more accurately describes what it does.
+ Make nice methods instead of poking in the structures.
+ Move some methods inside the relevant object source code.
+ Use GPtrArray to store objects instead of plain arrays, it is more
+ natural and allows us to more easily clean up.
+ Move the allocation of udp ports to the Stream object. The Stream object
+ contains the elements needed to stream the media to a client.
+ Improve the prepare and unprepare methods. Unprepare should now undo
+ everything prepare did. Improve also async unprepare when doing EOS on
+ shutdown. Make sure we always unprepare correctly.
+
+2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Unref server address clients connected to
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
+
+2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: don't ref server socket if it is NULL
+ Fixes test_bind_already_in_use unit test again after commit 6a497440.
+ https://bugzilla.gnome.org/show_bug.cgi?id=686644
+
+2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * tests/check/Makefile.am:
+ tests: Add libgio link dependency
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
+
+2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ rtsp-media-mapping: rename find_media vfunc to find_factory
+ The virtual method and class method should have the same name
+ so it is correctly represented in GIR file
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-server: fixed comments and GIR annotations
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
+
+2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: allow binding on port 0 (binds on a random port)
+
+2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: add bound-port property
+ bound-port can be used to retrieve the port number when the server is bound on
+ port 0, which binds on a random port.
+
+2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ rtsp-media-factory: make ::get_element overridable by GI bindings
+ The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
+ for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
+ as the invoker for ::get_element(), making it overridable by GI generated
+ bindings.
+
+2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: don't autoplug parsers in a loop
+ Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
+ h264parse forever.
+
+2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/Makefile.am:
+ Explicitly link against gio. Fix link error on mac.
+
+2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: add ttl to the transport header in SETUP
+ See https://bugzilla.gnome.org/show_bug.cgi?id=685561
+
+2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ client: Use client transport settings for multicast if allowed.
+ This patch makes it possible for the client to send transport settings for
+ multicast (destination && ttl). Client settings must be explicitly allowed or
+ the server will use its own settings.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
+
+2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6c0b52c to 6bb6951
+
+2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: do not destroy the rtsp watch
+ Don't destroy the client watch while dispatching. The rtsp watch is
+ automatically destroyed after the rtsp watch function closed() has
+ been called.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
+
+2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 4f962f7 to 6c0b52c
+
+2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix check for seekability
+
+2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use more GIO
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
+
+2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: remove obsolete includes
+
+2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
+ * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
+ be available in "on_new_ssrc". The transports are added in
+ gst_rtsp_media_set_state when going to PLAYING state. However,
+ "on_new_ssrc" might be called before this happens.
+ https://bugzilla.gnome.org/show_bug.cgi?id=683304
+
+2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: add signals for rtsp requests (fixes #683287)
+
+2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ add new-session signal to rtsp-client (fixes #683058)
+
+2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 668acee to 4f962f7
+
+2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-server: fixed segfault in gst_rtsp_server_create_socket
+ Do not assume that *error is set in g_socket_address_enumerator_next.
+ Added test_bind_already_in_use unit-test.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
+
+2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 94ccf4c to 668acee
+
+2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: make create_sdp virtual method
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
+
+2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 98e386f to 94ccf4c
+
+2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix docs
+
+2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: use an existing socket to establish HTTP tunnel
+ Make it possible to transfer a socket from an HTTP server to be used as
+ an RTSP over HTTP tunnel.
+
+2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp: Handle the blocksize parameter
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
+
+2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/rtspserver.c:
+ Have unit test get header from source dir, not installed dir
+ This makes compilation of unit tests work in a build directory other
+ than the source directory.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
+
+2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: update for gst_element_make_from_uri() changes
+
+2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * configure.ac:
+ * tests/Makefile.am:
+ * tests/check/Makefile.am:
+ * tests/check/gst/rtspserver.c:
+ rtsp: add unit test
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
+
+2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: don't collect media stats when going to NULL
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
+
+2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: don't leak transports
+
+2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: free transport on no_stream in SETUP handler
+
+2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: changed session media iteration
+ In client_unlink_session: now don't iterate in session->medias
+ list where items are removed by gst_rtsp_session_release_media.
+ Instead, repeatedly remove the first item.
+
+2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
+ GstRTSPSessionMedia is not a GObject type. When the
+ GstRTSPSession is freed, it will free the media.
+
+2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: plug pad leak in collect_streams
+ In gst_rtsp_media_factory_collect_streams: unref the srcpad that
+ was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
+ will take one reference, and the other reference will otherwise
+ give a memory leak.
+
+2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * configure.ac:
+ configure: suppress some warnings when debug is disabled
+ Warnings about unused variables should be suppressed if core has the
+ debug system disabled.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
+
+2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/libs/Makefile.am:
+ docs: fix build in uninstalled setup
+ Include gst-plugins-base libs properly.
+
+2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * docs/libs/gst-rtsp-server.types:
+ docs: include headers defining rtsp-server object types
+ Fixes compiler warnings during docs build.
+ https://bugzilla.gnome.org/show_bug.cgi?id=676824
+
+2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * configure.ac:
+ configure: Add warning flags for compiler when configuring
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
+
+2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 03a0e57 to 98e386f
+
+2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 1fab359 to 03a0e57
+
+2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix GSocketAddress leak in gst_rtsp_client_accept
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
+
+2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From f1b5a96 to 1fab359
+
+2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 92b7266 to f1b5a96
+
+2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From ec1c4a8 to 92b7266
+
+2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 3429ba6 to ec1c4a8
+
+2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp: fix compiler warnings
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
+
+2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From dc70203 to 3429ba6
+
+2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ rtsp-server: port to new thread API
+
+2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6db25be to dc70203
+
+2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-server: Fix compilation and compiler warnings
+
+2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * autogen.sh:
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ configure: Modernize autotools setup a bit
+ Also we now only create tar.bz2 and tar.xz tarballs.
+
+2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 464fe15 to 6db25be
+
+2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 7fda524 to 464fe15
+
+2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ * docs/libs/Makefile.am:
+ * docs/version.entities.in:
+ * gst-rtsp.spec.in:
+ * gst/rtsp-server/Makefile.am:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp-server.pc.in:
+ * tests/Makefile.am:
+ rtsp-server: Update versioning
+
+2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/0.10'
+ Conflicts:
+ gst/rtsp-server/rtsp-session-pool.c
+
+2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-server: Don't use deprecated GLib API
+
+2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Replace master with 0.11
+
+2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * docs/README:
+ A couple minor typo fixes
+
+2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix state of the appqueue
+
+2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory: use videoconvert
+
+2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory: change to new style caps
+
+2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-server: port to GIO
+ Port to GIO
+
+2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: fix build
+
+2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/README:
+ docs: fix for gst_rtsp_server_set_port() -> _set_service()
+ https://bugzilla.gnome.org/show_bug.cgi?id=666548
+
+2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * examples/Makefile.am:
+ First rule of gst-rtsp-server club: don't talk about gst-phonon
+
+2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/gst-rtsp-server.pc.in:
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp-server.pc.in:
+ pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
+ For consistency with all other modules.
+
+2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: update for new map API
+
+2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * .gitignore:
+ * bindings/Makefile.am:
+ * bindings/python/Makefile.am:
+ * bindings/python/arg-types.py:
+ * bindings/python/codegen/Makefile.am:
+ * bindings/python/codegen/__init__.py:
+ * bindings/python/codegen/argtypes.py:
+ * bindings/python/codegen/code-coverage.py:
+ * bindings/python/codegen/codegen.py:
+ * bindings/python/codegen/definitions.py:
+ * bindings/python/codegen/defsparser.py:
+ * bindings/python/codegen/docextract.py:
+ * bindings/python/codegen/docgen.py:
+ * bindings/python/codegen/fileprefix.override:
+ * bindings/python/codegen/fileprefixmodule.c:
+ * bindings/python/codegen/h2def.py:
+ * bindings/python/codegen/mergedefs.py:
+ * bindings/python/codegen/mkskel.py:
+ * bindings/python/codegen/override.py:
+ * bindings/python/codegen/reversewrapper.py:
+ * bindings/python/codegen/scmexpr.py:
+ * bindings/python/rtspserver-types.defs:
+ * bindings/python/rtspserver.defs:
+ * bindings/python/rtspserver.override:
+ * bindings/python/rtspservermodule.c:
+ * bindings/python/test.py:
+ * configure.ac:
+ python: remove pygst-based python bindings
+ pygi is the future, apparently.
+
+2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+
+ * common:
+ Automatic update of common submodule
+ From c463bc0 to 7fda524
+
+2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 2a59016 to c463bc0
+
+2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 0807187 to 2a59016
+
+2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 11f0cd5 to 0807187
+
+2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ example: update for new caps
+
+2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: port some more to 0.11
+ Fix caps.
+ Remove bufferlist stuff
+ Update for new API.
+ Add queue before appsink now that preroll-queue-len is gone.
+ Update for request pad changes.
+
+2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add a seekable boolean
+ Maintain the seekable state with a new variable instead of reusing the
+ is_live variable.
+
+2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Disallow seek in live media
+
+2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
+
+ * gst/rtsp-server/rtsp-server.c:
+ #ifdef statements for windows socket creation were missing
+
+2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From a39eb83 to 11f0cd5
+
+2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 605cd9a to a39eb83
+
+2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use method to access property
+
+2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add protocols property
+ Add a property to configure the allowed protocols in the media created from the
+ factory.
+
+2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add media-configure signal
+ Add signal to allow the application to configure the media after it was created
+ from the factory.
+
+2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use method to access property
+
+2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add protocols property
+ Add a property to configure the allowed protocols in the media created from the
+ factory.
+
+2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add media-configure signal
+ Add signal to allow the application to configure the media after it was created
+ from the factory.
+
+2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use media multicast group
+
+2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ retab some .h
+
+2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: copy and free the server ip address
+ Copy and free the server ip address to make memory management easier later.
+
+2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: configure multicast in media
+
+2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add property for multicast group
+ Add a property to configure the multicast group in the media.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add property for multicast group
+ Add a property to configure the multicast group in the media factory.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: do configuration of transport in one place
+ Move the configuration of the transport destination address to where we also
+ configure the other bits.
+
+2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use media multicast group
+
+2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ retab some .h
+
+2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: copy and free the server ip address
+ Copy and free the server ip address to make memory management easier later.
+
+2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: configure multicast in media
+
+2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add property for multicast group
+ Add a property to configure the multicast group in the media.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add property for multicast group
+ Add a property to configure the multicast group in the media factory.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: do configuration of transport in one place
+ Move the configuration of the transport destination address to where we also
+ configure the other bits.
+
+2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: destroy pipeline on client disconnect with no prior TEARDOWN.
+ The problem occurs when the client abruptly closes the connection without
+ issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
+ server is where the pipeline gets torn down. Since this handler is not called,
+ the pipeline remains and is up and running. Subsequent clients get their own
+ pipelines and if the do not issue TEARDOWNs then those pipelines will also
+ remain up and running. This is a resource leak.
+
+2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
+ For example, it can be used to retrieve source elements like appsrc, in a more
+ convenient way than subclassing get_element.
+
+2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: hold on to reference while using object
+
+2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: use new api
+
+2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: use unstable api
+
+2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix reference counting
+
+2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ fix compiler warnings about unused variables
+
+2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
+
+ * examples/test-launch.c:
+ * examples/test-readme.c:
+ * examples/test-uri.c:
+ * examples/test-video.c:
+ examples: tell rtsp uri when ready
+
+2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
+
+ * common:
+ Automatic update of common submodule
+ From 69b981f to 605cd9a
+
+2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: update for buffer API change
+
+2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile.am: 0.10 => @GST_MAJORMINOR@
+
+2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
+
+2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/.gitignore:
+ .gitignore: 0.10 => 0.11
+
+2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile.am: 0.10 => @GST_MAJORMINOR@
+
+2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 9e5bbd5 to 69b981f
+
+2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From fd35073 to 9e5bbd5
+
+2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 46dfcea to fd35073
+
+2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ media: port to new caps API
+
+2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ Updated Vala bindings.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Add a signal for newly connected clients.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/rtspserver.override:
+ python: override gst_rtsp_media_mapping_add_factory to fix refcounting
+
+2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-funnel.c:
+ * gst/rtsp-server/rtsp-funnel.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-server: port to 0.11
+
+2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * common:
+ add common
+
+2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ common
+ configure.ac
+
+2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From c3cafe1 to 46dfcea
+
+2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/Makefile.am:
+ * bindings/python/rtspserver.defs:
+ python bindings: wrap GstRTSPMediaFactoryClass vfuncs
+
+2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/arg-types.py:
+ python bindings: add GstRTSPUrlParam
+ Needed to implement MediaFactory virtual proxies
+
+2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/arg-types.py:
+ python bindings: fix returning GstRTSPUrl types
+
+2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/arg-types.py:
+ python bindings: add arg type for GstRTSPUrl
+
+2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/rtspserver.defs:
+ python bindings: fix the definition of MediaFactory.collect_stream
+
+2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 1ccbe09 to c3cafe1
+
+2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 193b717 to 1ccbe09
+
+2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From b77e2bf to 193b717
+
+2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * Makefile.am:
+ build: Include lcov.mak to allow test coverage report generation
+
+2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From d8814b6 to b77e2bf
+
+2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6aaa286 to d8814b6
+
+2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6aec6b9 to 6aaa286
+
+2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * autogen.sh:
+ autogen: wingo signed comment
+
+2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session: use full charset for RTSP session ID
+ As specified in RFC 2326 section 3.4 use full valid charset to make guessing
+ session ID more difficult.
+ https://bugzilla.gnome.org/show_bug.cgi?id=643812
+
+2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Don't install the funnel header
+
+2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 1de7f6a to 6aec6b9
+
+2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: require core/base 0.10.31
+ Needed at least for gst_plugin_feature_rank_compare_func().
+
+2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f94d739 to 1de7f6a
+
+2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: remove more unused code
+
+2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: remove duplicate filtering
+ Remove the duplicate filtering code now that we have a released -good version.
+ Give a warning instead.
+
+2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix default buffer size
+
+2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add property to configure the buffer-size
+ Add a property to configure the kernel UDP buffer size.
+
+2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add property to configure kernel buffer sizes
+ Add a property to configure the kernel UDP buffer size.
+
+2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: set PYGOBJECT_REQ before using it
+ https://bugzilla.gnome.org/show_bug.cgi?id=640641
+
+2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/Makefile.am:
+ docs: recursive into sub-directories on 'make upload'
+
+2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/version.entities.in:
+ docs: mention full version these docs are for, not just major-minor
+
+2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.8 ===
+
+2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.8
+
+2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: clarify docs a little
+
+2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: init debug category before starting thread
+
+2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: add realm to make it more spec compliant
+
+2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add locking
+
+2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ example: improve example docs a little
+
+2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: ensure the watch has a ref to the server
+
+2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: simpify channel function
+
+2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: simplify management of channel and source
+ We don't need to keep around the channel and source objects. Let the mainloop
+ and the source manage the source and channel respectively.
+
+2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * Makefile.am:
+ * configure.ac:
+ build tests
+
+2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/.gitignore:
+ * tests/Makefile.am:
+ * tests/test-cleanup.c:
+ tests: add tests directory and cleanup test
+
+2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ server: improve debugging in various objects
+
+2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: chain up to the parent finalize
+
+2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
+
+ * bindings/python/rtspserver-types.defs:
+ * bindings/python/rtspserver.defs:
+ * bindings/python/rtspserver.override:
+ * bindings/python/test.py:
+ gst-rtsp-server: update python bindings
+
+2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use the response from the clientstate
+ Create the response object only once and store in the client state.
+ Make all methods use the state response,
+
+2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: use signal to keep track of clients
+ Keep track of all the clients that the server creates and remove them when they
+ fire the 'closed' signal.
+
+2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: emit signal when closing
+
+2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-auth.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ media: enable per factory authorisations
+ Allow for adding a GstRTSPAuth on the factory and media level and check
+ permissions when accessing the factory.
+ Add hints to the auth methods for future more fine grained authorisation.
+ Add example application for per factory authentication.
+
+2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ rtsp-server: Pass ClientState structure arround
+ Pass the collected information for the ongoing request in a GstRTSPClientState
+ structure that we can then pass around to simplify the method arguments. This
+ will also be handy when we implement logging functionality.
+
+2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add methods to configure authorisation
+
+2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: unref auth in finalize
+
+2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: unref auth in finalize
+
+2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * docs/libs/gst-rtsp-server.types:
+ docs: add more docs
+
+2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: separate create and accept
+ Create separate create and accept methods so that subclasses can create custom
+ client object.
+ Configure the server in the client object and prepare for keeping track of
+ connected clients.
+
+2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add support for setting the server.
+ Add support for keeping a ref to the server that started this client
+ connection.
+
+2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: fix memleak and add some docs
+ Fix a memleak of the basic auth token.
+ Add docs for the helper function
+
+2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ client: delegate setup of auth to the manager
+ Delegate the configuration of the authentication tokens to the manager object
+ when configured.
+
+2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ auth: add authentication object
+ Add an object that can check the authorization of requests.
+ Implement basic authentication.
+ Add example authentication to test-video
+
+2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: move includes back
+ the includes are needed for sockaddr_in.
+
+2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp: move network includes where they are needed
+
+2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media.h: Minor corrections in comments.
+ Fixes #638944
+
+2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From e572c87 to f94d739
+
+2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * .gitignore:
+ * docs/.gitignore:
+ * docs/libs/.gitignore:
+ * examples/.gitignore:
+ * gst/rtsp-server/.gitignore:
+ gitignore: updates
+
+2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * docs/libs/Makefile.am:
+ docs: We don't build ps/pdf for API reference docs
+
+2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From ccbaa85 to e572c87
+
+2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 46445ad to ccbaa85
+
+2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/fs-funnel.c:
+ * gst/rtsp-server/fs-funnel.h:
+ * gst/rtsp-server/rtsp-funnel.c:
+ * gst/rtsp-server/rtsp-funnel.h:
+ * gst/rtsp-server/rtsp-media.c:
+ funnel: rename fsfunnel to rtspfunnel
+ Rename the funnel to avoid conflicts with the farsight one.
+
+2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/fs-funnel.c:
+ * gst/rtsp-server/fs-funnel.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: add and use fsfunnel
+ Add a copy of fsfunnel to the build because input-selector removed the (broken)
+ select-all property that we need.
+
+2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ gobject-introspection: use PKG_CONFIG_PATH specified at configure time
+ Use PKG_CONFIG_PATH specified at configure time (if any) as well
+ for the g-ir-compiler, rather than just assuming the env var has
+ been set.
+
+2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ * m4/Makefile.am:
+ * m4/codeset.m4:
+ build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
+
+2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ gobject-introspection: fix g-i build for uninstalled setup
+ Requires gst-plugins-base git (> 0.10.31.2).
+
+2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ examples: add some more options and comments
+
+2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: use right property type
+
+2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: attempt to configure buffer-lists
+ Attempt to configure buffer lists in the payloader for improved performance.
+
+2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: attempt to configure bigger UDP buffers
+ Attempt to configure bigger udp kernel send buffers to avoid overflowing the
+ send buffers with high bitrate streams.
+
+2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use the socket length from getsockname
+ Use the length returned by getsockname to perform the getnameinfo call because
+ the size can depend on the socket type and platform.
+ Fixes #638723
+
+2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: add uri factory to the docs
+
+2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.h:
+ docs: improve docs
+
+2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: add support for buffer lists
+ Add support for sending bufferlists received from appsink.
+ Fixes #635832
+
+2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ media: make method to retrieve the play range
+ Make a method to retrieve the playback range so that we can conditionally create
+ a different range for the SDP and the PLAY requests.
+
+2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal to notify of state changes
+
+2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ client: cleanup headers
+
+2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix typo
+
+2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ factory-uri: add support for gstpay
+ Add an option to prefer gstpay over decoder + raw payloader.
+
+2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ factory-uri: rework the autoplugger.
+ Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
+ before payloaders.
+
+2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: use better factory filter
+ Make better payloader filter based on autoplug rank and RTP use case.
+
+2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 169462a to 46445ad
+
+2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: set SO_REUSEADDR before bind
+ Set the SO_REUSEADDR _before_ bind() to make it actually work.
+
+2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: emit prepared signal when prepared
+ Make a 'prepared' signal and emit it when we successfully prepared the element.
+ This signal can be used to configure the media object after it has been prepared
+ for streaming.
+
+2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 011bcc8 to 169462a
+
+2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
+
+ python an optional dependency
+ * configure.ac: Move up valgrind and g-i checks. Make the python
+ dependency optional, as it was before.
+
+2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ common
+ configure.ac
+
+2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: update range when active clients changed
+ When we changed the number of active clients, update the current range
+ information because we want the second client connecting to a shared resource
+ continue from where the stream currently.
+
+2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ factory-uri: add colorspace and fix pt
+ Rework the way we pass data to the autoplugger.
+ When we have raw caps, plug a converter element to make pluggin to raw
+ payloaders more successful.
+ Make sure all dynamically plugged payloaders have a unique payload types.
+
+2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-uri.c:
+ example: add example of the uri factory
+
+2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-server.h:
+ factory-uri: add a factory to stream any URI
+ Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
+ when we have one.
+
+2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: ignore spurious ASYNC_DONE messages
+ When we are dynamically adding pads, the addition of the udpsrc elements will
+ trigger an ASYNC_DONE. We have to ignore this because we only want to react to
+ the real ASYNC_DONE when everything is prerolled.
+
+2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: make lock macro
+
+2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-server: Remove unused variable and dead assignment
+
+2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-readme.c:
+ * examples/test-sdp.c:
+ * examples/test-video.c:
+ examples: Run gst-indent
+
+2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-server: Run gst-indent
+ Since it wasn't using the upstream common previously, there was no
+ indentation check before commiting.
+
+2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: Some more doc fixups
+
+2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * Makefile.am:
+ Makefile: Add cruft-cleaning support
+
+2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * Makefile.am:
+ * configure.ac:
+ * docs/Makefile.am:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * docs/libs/gst-rtsp-server.types:
+ * docs/version.entities.in:
+ docs: Add gtk-doc build system
+
+2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile.am: Use standard GIR make behaviour
+
+2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * autogen.sh:
+ * configure.ac:
+ autogen/configure: Bring more in sync to standard gst module behaviour
+
+2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: warn and fail when gstrtpbin is not found
+
+2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: open 0.11 branch
+
+2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * .gitmodules:
+ * common:
+ Add common submodule
+
+2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common/ChangeLog:
+ * common/Makefile.am:
+ * common/c-to-xml.py:
+ * common/check.mak:
+ * common/coverage/coverage-report-entry.pl:
+ * common/coverage/coverage-report.pl:
+ * common/coverage/coverage-report.xsl:
+ * common/coverage/lcov.mak:
+ * common/gettext.patch:
+ * common/glib-gen.mak:
+ * common/gst-autogen.sh:
+ * common/gst-xmlinspect.py:
+ * common/gst.supp:
+ * common/gstdoc-scangobj:
+ * common/gtk-doc-plugins.mak:
+ * common/gtk-doc.mak:
+ * common/m4/.gitignore:
+ * common/m4/Makefile.am:
+ * common/m4/README:
+ * common/m4/as-ac-expand.m4:
+ * common/m4/as-auto-alt.m4:
+ * common/m4/as-compiler-flag.m4:
+ * common/m4/as-compiler.m4:
+ * common/m4/as-docbook.m4:
+ * common/m4/as-libtool-tags.m4:
+ * common/m4/as-libtool.m4:
+ * common/m4/as-python.m4:
+ * common/m4/as-scrub-include.m4:
+ * common/m4/as-version.m4:
+ * common/m4/ax_create_stdint_h.m4:
+ * common/m4/check.m4:
+ * common/m4/glib-gettext.m4:
+ * common/m4/gst-arch.m4:
+ * common/m4/gst-args.m4:
+ * common/m4/gst-check.m4:
+ * common/m4/gst-debuginfo.m4:
+ * common/m4/gst-default.m4:
+ * common/m4/gst-doc.m4:
+ * common/m4/gst-error.m4:
+ * common/m4/gst-feature.m4:
+ * common/m4/gst-function.m4:
+ * common/m4/gst-gettext.m4:
+ * common/m4/gst-glib2.m4:
+ * common/m4/gst-libxml2.m4:
+ * common/m4/gst-plugindir.m4:
+ * common/m4/gst-valgrind.m4:
+ * common/m4/gtk-doc.m4:
+ * common/m4/introspection.m4:
+ * common/m4/pkg.m4:
+ * common/mangle-tmpl.py:
+ * common/plugins.xsl:
+ * common/po.mak:
+ * common/release.mak:
+ * common/scangobj-merge.py:
+ * common/upload.mak:
+ common: Remove static version
+
+2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
+
+ * common/m4/introspection.m4:
+ Update introspection.m4 to match usage
+
+2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * README:
+ README: update
+ Remove old stuff from the README
+
+2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.7 ===
+
+2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.7
+
+2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-ogg.c:
+ test-ogg: remove parsers
+ Remove the parsers, they are not needed anymore as oggdemux now outputs normal
+ buffers with timestamps. Using the parsers also seems to break things.
+
+2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated Vala bindings
+
+2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * common/m4/introspection.m4:
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ Added initial gobject-introspection support
+
+2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: don't use host for shared hash key
+ When we generate the key to share made between connections, don't include the
+ host used to connect so that we can share media even if between clients that
+ connected with localhost and ones with the ip address.
+
+2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * bindings/vala/Makefile.am:
+ build: fix distcheck
+
+2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Update Vala bindings
+
+2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * bindings/vala/Makefile.am:
+ * configure.ac:
+ Fix configure checks and installation location for Vala bindings
+ Fixes bug #628676.
+
+2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.6 ===
+
+2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: release 0.10.6
+
+2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: help the compiler a little
+
+2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ media: cleanup media transport before freeing
+ Cleanup the media transport data before freeing. In particular, remove the qdata
+ from the rtpsource object.
+
+2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media-factory: add eos-shutdown property
+ Add an eos-shutdown property that will send an EOS to the pipeline before
+ shutting it down. This allows for nice cleanup in case of a muxer.
+ Fixes #625597
+
+2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: use multiudpsink send-duplicates when we can
+ If we have a new enough multiudpsink with the send-duplicates property, use this
+ instead of doing our own filtering. Our custom filtering code should eventually
+ be removed when we can depend on a released -good.
+
+2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't leak destinations
+ Refactor and cleanup the destinations array when the stream is destroyed.
+
+2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: don't add udp addresses multiple times
+ Keep track of the udp addresses we added to udpsink and never add the same udp
+ destination twice. This avoids duplicate packets when using multicast.
+
+2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: disable use of SO_LINGER
+ SO_LINGER cause the client to fail to receive a TEARDOWN message because the
+ server close()s the connection.
+
+2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: use 5 second linger period in SO_LINGER
+ Wait 5 seconds before clearing the send buffers and reseting the connection with
+ the client when we do a close. This should be enough time to get the message to
+ the client.
+ See #622757
+
+2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: use SO_LINGER
+ SO_LINGER on the socket will make sure that any pending data on the socket is
+ flushed ASAP and that the socket connection is reset. This makes sure that the
+ socket can be reused immediately.
+ Fixes 622757
+
+2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ README: add blurb about shared media factories
+
+2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Add stdlib.h for atoi()
+
+2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * bindings/python/Makefile.am:
+ * bindings/vala/Makefile.am:
+ build: distcheck fixes
+ Fix 'make distcheck', somewhat (it still fails because it tries to
+ install files into /usr/share/vala/vapi/ irrespective of the
+ configured prefix).
+
+2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump core/base requirements to released version
+ Makes things less confusing for people.
+
+2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: fail if GStreamer core/base requirements are not met
+
+2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve client cleanups
+ Make sure the session does not timeout when using TCP. We need to do this
+ because quicktime player does not send RTCP for some reason in tunneled
+ mode.
+ Refactor some cleanup code.
+ Fixes #612915
+
+2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: add support for prevent session timeouts
+ Add an atomix counter to prevent session timeouts when we are, for example,
+ streaming over TCP.
+
+2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix unlink on session timeouts
+ When our session times out, make sure we unlink all streams in this
+ session.
+ Remove the tunnelid when closing the connection.
+
+2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: small cleanups
+
+2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: handle lost_tunnel callbacks
+ Handle lost_tunnel callbacks and use it to store the tunnelid back into the
+ hashtable so that we can reuse it for when the client reopens the POST
+ socket.
+ Close the connection after a TEARDOWN.
+ Make sure or watchid is cleared when the watch is removed.
+ Fixes #612915
+
+2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-server: add more support for multicast
+
+2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: allow configuration of allowed lower transport
+
+2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp: keep track of server ip and ipv6
+ Keep track of how the client connected to the server and setup the udp ports
+ with the same protocol.
+ Copy the server ip address in the SDP so that clients can send RTCP back to
+ us.
+
+2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: indent
+
+2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use right size for malloc
+
+2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: comment ipv6 server listening address
+
+2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: allow for ipv6 sockets
+
+2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: rework server part
+ Allow setting a bind address, make sure we can deal with ipv6.
+ Remove the port property and change with the service property.
+
+2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.h:
+ media: update comments a little
+
+2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: make content-base better
+ Use the URI formatting functions to make a content-base. Also make sure that
+ there is a trailing / at the end.
+
+2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: guard against invalid paths
+
+2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ test: catch server bind errors
+
+2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtspmedia: emit "unprepared" if _prepare fails.
+ Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
+ media object is removed from its factory's cache.
+
+2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: collect media position when seek completes
+
+2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: call unlink_streams in client finalize
+ Fixes #599027
+
+2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: limit the time to wait to something huge
+ Avoid waiting forever but limit the timeout to 20 seconds.
+
+2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: reindent and check for prepared status
+
+2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ media: avoid doing _get_state() for state changes
+ When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
+ until the media is prerolled or in error. This avoids doing a blocking call of
+ gst_element_get_state() that can cause lockups when there is an error.
+ Fixes #611899
+
+2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: reindent
+
+2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: better error handling
+ Improve the error handling a bit.
+
+2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: rework transport parsing
+ Rework the transport parsing code so that we can ignore transports we don't
+ support instead of just picking the first one we can parse.
+ Configure a (for now hardcoded) destination for multicast transports.
+
+2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: set multicast sink parameters
+ Disable loop and automatic multicast join on the udpsink elements.
+ Add some more debug info.
+ Reset some state variables in the right place.
+ Use the right port numbers for multicast.
+
+2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: handle transport setup correctly
+ Handle UDP, MCAST and TCP transport negotiation more correctly.
+ Store the server session SSRC in the transport.
+
+2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: implement error_full
+ Implement error_full to avoid some segfaults when the rtspconnection calls it.
+ See #608245
+
+2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ docs: update docs and comments
+
+2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: make server work better when behind a proxy
+
+2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
+
+2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ Use GStreamer's debugging subsystem
+
+2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
+
+2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.5 ===
+
+2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.5
+
+2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump required versions
+
+2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: call weak-unref on client->sessions from finalize
+ Fixes bug #596305
+
+2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Fixed crasher where caps got unref'ed too often
+
+2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * configure.ac:
+ * pkgconfig/.gitignore:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
+ Added pkg-config file to use gst-rtsp-server uninstalled
+
+2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: add some docs
+
+2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp: Use gst_rtsp_watch_send_message().
+ Use gst_rtsp_watch_send_message() since the old API which used
+ gst_rtsp_watch_queue_message() has been deprecated.
+
+2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.4 ===
+
+2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ Release 0.10.4
+
+2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp: allocate channels in TCP mode
+ When the client does not provide us with channels in TCP mode, allocate channels
+ ourselves.
+
+2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: don't crash when tunnelid is missing
+ When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
+ don't crash but return an error response to the client.
+ Fixes #589489
+
+2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ bindings: update vala bindings with new method
+
+2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ sessionpool: add function to filter sessions
+ Add generic function to retrieve/remove sessions.
+
+2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump core/base requirements to release
+
+2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix indentation
+
+2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
+
+2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ set state and remove elements of media in for loop
+
+2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ Added gst_rtsp_media_remove_elements function to Vala bindings
+
+2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Added gst_rtsp_media_remove_elements function
+
+2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Don't use name for gstrtpbin so we can add multiple instances to the pipeline
+
+2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated Vala bindings
+
+2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Added vmethod unprepare to GstRTSPMedia
+ The default implementation sets the state of the pipeline to GST_STATE_NULL
+
+2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ Made collect_streams function public
+
+2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ Added vmethod create_pipeline to GstRTSPMediaFactory
+ The pipeline is created in this method and the GstRTSPMedia's element is added to it
+
+2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use g_source_destroy()
+ We need to use g_source_destroy() because we might have added the source to a
+ different main context than the default one.
+
+2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ rtsp: prepare for handling GET/SET_PARAMETER
+ Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
+ is a body now.
+ Fix return codes of handlers.
+
+2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't leak session pads
+
+2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: clean up the messages a bit
+
+2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: warn and skip streams without media
+
+2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ vala: Fixed typo in header file of RTSPMediaStream
+
+2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix message
+ Fix a debug message
+ Make dumping RTCP stats configurable
+
+2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: be less verbose and leak less
+
+2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't leak the destination address
+
+2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp: use RTCP to keep the session alive
+ Use the RTCP rtcp-from stats field to find the associated session and use this
+ to keep the session alive.
+
+2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: add 5sec to the real session timeout
+ Allow the session to live 5sec longer before really timing out. This should give
+ clients some extra time to keep the session active.
+
+2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: replay OK to GET/SET_PARAMETER
+ Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
+ so that we return OK for those requests.
+
+2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: keep track of active transports
+ Keep track of which transport is active to avoid closing the connection too
+ soon.
+ Remove the destination transport also when going to NULL.
+ Print some stats about the SDES and other RTCP messages we receive from the
+ clients.
+
+2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-sdp.c:
+ example: add SDP relay example
+
+2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: also count active TCP connections
+
+2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp: add support for dynamic elements
+ Add support for dynamic elements.
+ Don't set live pipelines back to paused.
+
+2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: don't add encoding name when absent in caps
+
+2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: warn when we can't do RTP-Info
+
+2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: factor out the stream construction
+
+2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: only add RTP-Info when we have the info
+ Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
+ depayloader.
+
+2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.3 ===
+
+2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release: 0.10.3
+ - Fixes a bug where it put the wrong verion in pkgconfig
+ - Link RTP and RTCP sources
+
+2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: link the RTP udpsrc to the session manager
+ Link the RTP udpsrc and the appsrc to the session manager so that they don't
+ shut down when the client sends a packet to open firewalls.
+
+2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * pkgconfig/gst-rtsp-server.pc.in:
+ Don't use hard-coded version number in pkg-config file
+
+2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.2 ===
+
+2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.2
+
+2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * .gitignore:
+ * common/m4/.gitignore:
+ * examples/.gitignore:
+ * pkgconfig/.gitignore:
+ add some .gitignore files
+
+2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: seek to key frames
+
+2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: emit the unprepared signal by id
+ Emit the unprepared signal by id instead of name and set the media as
+ reused.
+
+2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
+
+2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Added finalize function to GstRTPSPServer to unref session pool and media mapping
+
+2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated vala bindings
+
+2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ server: use appsink and appsrc with the API
+ Use the appsink/appsrc API instead of the signals for higher
+ performance.
+
+2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-ogg.c:
+ tests: set the payload type correctly
+
+2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: connect to the unprepare signal
+ Connect to the unprepare signal for non-reusable media so that we can remove
+ them from the cache.
+
+2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal to notify of unprepare
+
+2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: more work on making the media shared
+ Add a reusable flag to medias, indicating that they can be reused after a state
+ change to NULL.
+ Small cleanups.
+
+2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-readme.c:
+ examples: mark the example as shared for testing
+
+2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ client: support shared media
+ Always perform the state actions even if the target state of the pipeline is
+ already correct, we still want to add/remove the transports when we are dealing
+ with shared media.
+ Keep a counter of the number of active transports for a media so that we can use
+ this to perform a state change when needed.
+ Perform a state change of the pipeline only when the first transport was added
+ or when there are no active transports.
+
+2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix refcounting crasher
+ Don't need to remove the weak refs in the finalize methods, they are already
+ removed in the dispose.
+ Don't register the callback with a DestroyNofity.
+
+2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Fix rtsp client refcount management in TCP mode.
+ Don't unref a client ref we never had. Fixes an unref
+ of an already-free client object after a client
+ teardown request for me.
+
+2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ docs: fix typo in API docs
+
+2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ More seeking fixes.
+ Keep the udp sources in playing even if we go to paused. unlock the sources when
+ we shut down.
+ Add some more debug info.
+ Only seek when we need to.
+ Keep track of the position when we go to paused.
+
+2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add beginnings of seeking.
+ Parse the Range header and perform a seek on the pipeline for the requested
+ position. It's disabled currently until I figure out what's going wrong.
+
+2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ allow pause requests for now.
+ --
+
+2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Remove weak ref on the session in teardown
+ We need to remove our weakref from the session when we do a teardown because
+ else we close the TCP connection prematurely.
+
+2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Do some more session cleanup
+ Make session timeout kill the TCP connection that currently watches the
+ session.
+ Remove the client timeout property.
+
+2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add TCP transports
+ Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
+ connection.
+
+2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-launch.c:
+ Add example server that takes launch lines
+ Add an example server that streams any -launch line.
+
+2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-readme.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add support for live streams
+ Add support for live streams and ranges
+ Start on handling TCP data transfer.
+
+2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Free the pipeline before other things
+ ---
+
+2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Only free the pending tunnel if there is one
+ --
+
+2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-server: Add support for tunneling
+ Add support for tunneling over HTTP.
+ Use new connection methods to retrieve the url.
+ Dispatch messages based on the message type instead of blindly
+ assuming it's always a request.
+ Keep track of the watch id so that we can remove it later.
+ Set the media pipeline to NULL before unreffing the pipeline.
+
+2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Fix for channel -> watch rename in gstreamer
+ Rename the RTSPChannel to RTSPWatch and remove an unused variable.
+
+2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Use ASYNC RTSP io
+ Use the async RTSP channels instead of spawning a new thread for each client.
+ If a sessionid is specified in a request, fail if we don't have the session.
+
+2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Add better debug info
+ Add some better debug info.
+
+2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ Time out sessions
+ Add support for session timeouts in the example.
+
+2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ Pass GTimeVal around for performance reasons
+ Get the current time only once and pass it around so that sessions don't have to
+ get the current time anymore.
+ Add experimental support for a GSource that dispatches when the session needs to
+ be cleaned up.
+
+2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add better support for session timeouts
+ Add a method to request the number of milliseconds when a session will timeout.
+
+2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add suport for RTP manager monitoring
+ Add the first stage in monitoring the rtp manager.
+ Make sure we don't update the state to something we don't want.
+
+2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Add support for session keepalive
+ Get and update the session timeout for all requests. get the session as early as
+ possible.
+
+2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Handle media bus messages
+ Handle media bus messages in a custom mainloop and dispatch them to the
+ RTSPMedia objects. Let the default implementation handle some common messages.
+
+2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ Some more session timeout handling
+ Move the session header setting code to a central place so that we always add
+ the timeout parameter too.
+ Handle timeouts by running the session cleanup code.
+ Stop media before cleaning up.
+
+2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Add timeout property
+ Add a timeout property ot the client and make the other properties into GObject
+ properties.
+
+2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Use getters and setters in property code
+ Use the getters and setters for the timeout property instead of locking
+ ourselves.
+
+2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
+
+2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add more timeout stuff
+ Add method to check if a session is expired.
+ Add method to perform cleanup on a session pool.
+
+2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add beginnings of session timeouts and limits
+ Add the timeout value to the Session header for unusual timeout values.
+ Allow us to configure a limit to the amount of active sessions in a pool. Set a
+ limit on the amount of retry we do after a sessionid collision.
+ Add properties to the sessionid and the timeout of a session. Keep track of
+ creation time and last access time for sessions.
+
+2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Cleanup of sessions and more
+ Fix the refcounting of media and sessions in the client. Properly clean up the
+ session data when the client performs a teardown.
+ Add Server header to responses.
+ Allow for multiple uri setups in one session.
+ Add Range header to the PLAY response and add the range attribute to the SDP
+ message.
+ Fix the session pool remove method, it used the wrong key in the hashtable. Also
+ give the ownership of the sessionid to the session object.
+
+2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Rename a variable
+ Rename the 'server_port' variable to simply 'port'.
+
+2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Rework the way we handle transports for streams
+ Make the media accept an array of transports for the streams that we have
+ configured for the play/pause requests.
+ Implement server states for a client and its media.
+ Require 0.10.22.1 (git HEAD) of gstreamer.
+
+2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ Drop const from functions dealing with urls
+ Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
+ have the right const in them.
+
+2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ Fix various leaks
+ Fix some leaks.
+
+2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ More cleanups
+ Don't keep a reference to the GstRTSPMedia in the stream.
+ Free more things when freeing the GstRTSPMedia.
+
+2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ More docs and small cleanups
+ Add some more docs and update the README
+ Cleanup some method names.
+ Remove an unneeded idx field in the GstRTSPMediaStream
+
+2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * examples/Makefile.am:
+ * examples/test-readme.c:
+ Add a README and more example code
+ Add a README file that contains a small introduction on how to use the server
+ along with the example code explained in the readme.
+
+2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ Fix some leaks and change default port
+ Fix some memory leaks by setting the udpsrc elements to the unlocked state after
+ we finished the initial preroll. If we keep them locked, setting the pipeline to
+ NULL will not stop and clean up the sources correctly.
+ Change the default RTSP port to 8554 aka the official alternative RTSP port.
+
+2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Cleanups to the session object
+ Remove some unneeded variables in the session state of a stream such as the
+ owner media and the server transport.
+ Get the configuration of a media stream in a session based on the media_stream
+ in the original object instead of our cached index.
+ Free more data in the finalize method.
+
+2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Cleanups and reuse media from DESCRIBE
+ Handle thread create errors.
+ Rename some internal methods to better match what they actually do.
+ Handle misconfiguration of session_pool and media_mapping gracefully.
+ Cache the DESCRIBE media and uri in the client connection and reuse them when
+ we receive a SETUP request in the same connection for the same uri.
+ Cleanup the client connection object.
+
+2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add shared properties to media and factory
+ Add the shared property to media.
+ Implement some simple caching in the factory depending on if the media is shared
+ or not.
+
+2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Add a little comment
+ Add some comment about the content-base header.
+
+2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/main.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Reorganize things, prepare for media sharing
+ Added various other test server examples
+ Move the SDP message generation to a separate helper.
+ Refactor common code for finding the session.
+ Add content-base for realplayer compatibility
+ Clean up request uris before processing for better vlc compatibility.
+ Move prerolling and pipeline construction to the RTSPMedia object.
+ Use multiudpsink for future pipeline reuse.
+
+2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ Back to development
+ Back to 0.10.1.1
+
+=== release 0.10.1 ===
+
+2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ Make 0.10.1 release
+ Release 0.10.1
+
+2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * bindings/vala/Makefile.am:
+ Fix make dist
+ Add more directories and files to the dist.
+
+2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/python/Makefile.am:
+ * bindings/python/rtspserver.override:
+ Fixed compile error of python bindings
+
+2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Marked values as nullable accordingly
+
+2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated Vala bindings
+
+2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ Cleanups and doc updates
+ Add some more documentation and do some minor cleanups here and there.
+
+2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ More improvements
+ Rename GstRTSPMediaBin to GstRTSPMedia
+ Parse the request url into a GstRTSPUri object and pass this object to the
+ various handlers and methods that require the uri.
+
+2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/main.c:
+ Update example
+ Add some more docs and remove some old code from the example.
+
+2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Handle state change failures better
+ Handle state change failures better when changing the state of the pipeline to
+ determine the SDP.
+
+2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ Make element creation more extendible
+ Add get_element vmethod to the default MediaFactory so that subclasses can just
+ override that method and still use the default logic for making a MediaBin from
+ that.
+
+2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/main.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Make the server handle arbitrary pipelines
+ Make GstMediaFactory an object that can instantiate GstMediaBin objects.
+ The GstMediaBin object has a handle to a bin with elements and to a list of
+ GstMediaStream objects that this bin produces.
+ Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
+ with methods to register and remove those mappings.
+ Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
+ used by the server instance.
+ Modify the example application so that it shows how to create custom pipelines
+ attached to a specific mount point.
+ Various misc cleanps.
+
+2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Allow setting a custom media factory for a server
+
+2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Allow setting a custom media factory for a client.
+
+2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Add Makefile entry for the media factory
+
+2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ Add media factory to map urls to media pipeline objects.
+
+2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add comments. Remove unused field
+
+2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ Allow custom session pools to override the session id allocation algorithms Add some comments.
+
+2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.h:
+ Add some comments.
+
+2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Move the connection code in one place Add some comments
+
+2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Make vmethod to create and accept new clients. Add some docs.
+
+2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
+
+2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Name the parameters more appropriately.
+
+2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Do some more cleanup of the session pool.
+
+2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ Check if return value of gst_rtsp_session_get_media is not NULL
+
+2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Install rtsp-session and rtsp-session-pool headers
+
+2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * .gitignore:
+ * Makefile.am:
+ * acinclude.m4:
+ * bindings/python/Makefile.am:
+ * bindings/python/arg-types.py:
+ * bindings/python/codegen/Makefile.am:
+ * bindings/python/codegen/__init__.py:
+ * bindings/python/codegen/argtypes.py:
+ * bindings/python/codegen/code-coverage.py:
+ * bindings/python/codegen/codegen.py:
+ * bindings/python/codegen/definitions.py:
+ * bindings/python/codegen/defsparser.py:
+ * bindings/python/codegen/docextract.py:
+ * bindings/python/codegen/docgen.py:
+ * bindings/python/codegen/fileprefix.override:
+ * bindings/python/codegen/fileprefixmodule.c:
+ * bindings/python/codegen/h2def.py:
+ * bindings/python/codegen/mergedefs.py:
+ * bindings/python/codegen/mkskel.py:
+ * bindings/python/codegen/override.py:
+ * bindings/python/codegen/reversewrapper.py:
+ * bindings/python/codegen/scmexpr.py:
+ * bindings/python/rtspserver-types.defs:
+ * bindings/python/rtspserver.defs:
+ * bindings/python/rtspserver.override:
+ * bindings/python/rtspservermodule.c:
+ * configure.ac:
+ Add python bindings.
+
+2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * bindings/Makefile.am:
+ * configure.ac:
+ Don't go into python dir when requirements for python bindings are missing
+
+2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * bindings/Makefile.am:
+ * bindings/vala/Makefile.am:
+ * configure.ac:
+ Install Vala bindings if vala is available
+
+2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.deps:
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/gst-rtsp-server.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.deps:
+ * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
+ * bindings/vala/packages/gst-rtsp-server-0.10.files:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
+ * bindings/vala/packages/gst-rtsp-server.deps:
+ * bindings/vala/packages/gst-rtsp-server.excludes:
+ * bindings/vala/packages/gst-rtsp-server.files:
+ * bindings/vala/packages/gst-rtsp-server.gi:
+ * bindings/vala/packages/gst-rtsp-server.metadata:
+ * bindings/vala/packages/gst-rtsp-server.namespace:
+ Regenerated Vala bindings
+
+2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server.vapi:
+ * bindings/vala/packages/gst-rtsp-server.metadata:
+ Fixed typo in included headers for vala bindings
+
+2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * Makefile.am:
+ * configure.ac:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gst-rtsp-server.pc.in:
+ Added pkgconfig file
+
+2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server.vapi:
+ * bindings/vala/packages/gst-rtsp-server.excludes:
+ * bindings/vala/packages/gst-rtsp-server.gi:
+ * bindings/vala/packages/gst-rtsp-server.metadata:
+ Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
+
+2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server.vapi:
+ * bindings/vala/packages/gst-rtsp-server.deps:
+ * bindings/vala/packages/gst-rtsp-server.files:
+ * bindings/vala/packages/gst-rtsp-server.gi:
+ * bindings/vala/packages/gst-rtsp-server.metadata:
+ * bindings/vala/packages/gst-rtsp-server.namespace:
+ Added Vala bindings
+
+2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
+
+2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
+
+ * examples/Makefile.am:
+ * gst/rtsp-server/Makefile.am:
+ Put GStreamer version in library name
+
+2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * gst/rtsp-server/Makefile.am:
+ Fix some issues to pass distcheck
+
+2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Added port property to GstRTSPServer class.
+
+2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * Makefile.am:
+ * autogen.sh:
+ * configure.ac:
+ * examples/Makefile.am:
+ * examples/main.c:
+ * gst/Makefile.am:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * src/Makefile.am:
+ * src/main.c:
+ * src/rtsp-client.c:
+ * src/rtsp-client.h:
+ * src/rtsp-media.c:
+ * src/rtsp-media.h:
+ * src/rtsp-server.c:
+ * src/rtsp-server.h:
+ * src/rtsp-session-pool.c:
+ * src/rtsp-session-pool.h:
+ * src/rtsp-session.c:
+ * src/rtsp-session.h:
+ Split in library and example program
+
+2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
+
+ * src/rtsp-client.h:
+ Removed obsolete variable
+
+2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
+
+ * src/rtsp-client.c:
+ * src/rtsp-client.h:
+ Removed pipeline variable GstRTSPClient, because it's only used in one function
+
+2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * src/rtsp-media.c:
+ Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
+
+2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
+
+ * src/rtsp-session.c:
+ Initialize some more vars.
+
+2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
+
+ * src/rtsp-session.c:
+ Initialize variable to avoid compiler warning.
+
+2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
+
+ * .gitignore:
+ Add a reasonable generic .gitignore
+
--- /dev/null
+Installation Instructions
+*************************
+
+Copyright (C) 1994-1996, 1999-2002, 2004-2013 Free Software Foundation,
+Inc.
+
+ Copying and distribution of this file, with or without modification,
+are permitted in any medium without royalty provided the copyright
+notice and this notice are preserved. This file is offered as-is,
+without warranty of any kind.
+
+Basic Installation
+==================
+
+ Briefly, the shell command `./configure && make && make install'
+should configure, build, and install this package. The following
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+instructions specific to this package. Some packages provide this
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+in *note Makefile Conventions: (standards)Makefile Conventions.
+
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+you want to change it or regenerate `configure' using a newer version
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+ The simplest way to compile this package is:
+
+ 1. `cd' to the directory containing the package's source code and type
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+
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+ some messages telling which features it is checking for.
+
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+
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+
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+
+Compilers and Options
+=====================
+
+ Some systems require unusual options for compilation or linking that
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+
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+is an example:
+
+ ./configure CC=c99 CFLAGS=-g LIBS=-lposix
+
+ *Note Defining Variables::, for more details.
+
+Compiling For Multiple Architectures
+====================================
+
+ You can compile the package for more than one kind of computer at the
+same time, by placing the object files for each architecture in their
+own directory. To do this, you can use GNU `make'. `cd' to the
+directory where you want the object files and executables to go and run
+the `configure' script. `configure' automatically checks for the
+source code in the directory that `configure' is in and in `..'. This
+is known as a "VPATH" build.
+
+ With a non-GNU `make', it is safer to compile the package for one
+architecture at a time in the source code directory. After you have
+installed the package for one architecture, use `make distclean' before
+reconfiguring for another architecture.
+
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+compiler but only a single `-arch' option to the preprocessor. Like
+this:
+
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+ CXX="g++ -arch i386 -arch x86_64 -arch ppc -arch ppc64" \
+ CPP="gcc -E" CXXCPP="g++ -E"
+
+ This is not guaranteed to produce working output in all cases, you
+may have to build one architecture at a time and combine the results
+using the `lipo' tool if you have problems.
+
+Installation Names
+==================
+
+ By default, `make install' installs the package's commands under
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+
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+
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+
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+prefix=/alternate/directory' will choose an alternate location for all
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+but not in terms of `${prefix}', must each be overridden at install
+time for the entire installation to be relocated. The approach of
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+the GNU Coding Standards, and ideally causes no recompilation.
+However, some platforms have known limitations with the semantics of
+shared libraries that end up requiring recompilation when using this
+method, particularly noticeable in packages that use GNU Libtool.
+
+ The second method involves providing the `DESTDIR' variable. For
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+it does better at avoiding recompilation issues, and works well even
+when some directory options were not specified in terms of `${prefix}'
+at `configure' time.
+
+Optional Features
+=================
+
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+
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+
+Particular systems
+==================
+
+ On HP-UX, the default C compiler is not ANSI C compatible. If GNU
+CC is not installed, it is recommended to use the following options in
+order to use an ANSI C compiler:
+
+ ./configure CC="cc -Ae -D_XOPEN_SOURCE=500"
+
+and if that doesn't work, install pre-built binaries of GCC for HP-UX.
+
+ HP-UX `make' updates targets which have the same time stamps as
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+
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+parse its `<wchar.h>' header file. The option `-nodtk' can be used as
+a workaround. If GNU CC is not installed, it is therefore recommended
+to try
+
+ ./configure CC="cc"
+
+and if that doesn't work, try
+
+ ./configure CC="cc -nodtk"
+
+ On Solaris, don't put `/usr/ucb' early in your `PATH'. This
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+
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+not `/usr/local'. It is recommended to use the following options:
+
+ ./configure --prefix=/boot/common
+
+Specifying the System Type
+==========================
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+
+ CPU-COMPANY-SYSTEM
+
+where SYSTEM can have one of these forms:
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+ KERNEL-OS
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+
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+platform different from the build platform, you should specify the
+"host" platform (i.e., that on which the generated programs will
+eventually be run) with `--host=TYPE'.
+
+Sharing Defaults
+================
+
+ If you want to set default values for `configure' scripts to share,
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+`PREFIX/etc/config.site' if it exists. Or, you can set the
+`CONFIG_SITE' environment variable to the location of the site script.
+A warning: not all `configure' scripts look for a site script.
+
+Defining Variables
+==================
+
+ Variables not defined in a site shell script can be set in the
+environment passed to `configure'. However, some packages may run
+configure again during the build, and the customized values of these
+variables may be lost. In order to avoid this problem, you should set
+them in the `configure' command line, using `VAR=value'. For example:
+
+ ./configure CC=/usr/local2/bin/gcc
+
+causes the specified `gcc' to be used as the C compiler (unless it is
+overridden in the site shell script).
+
+Unfortunately, this technique does not work for `CONFIG_SHELL' due to
+an Autoconf limitation. Until the limitation is lifted, you can use
+this workaround:
+
+ CONFIG_SHELL=/bin/bash ./configure CONFIG_SHELL=/bin/bash
+
+`configure' Invocation
+======================
+
+ `configure' recognizes the following options to control how it
+operates.
+
+`--help'
+`-h'
+ Print a summary of all of the options to `configure', and exit.
+
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+
+`--prefix=DIR'
+ Use DIR as the installation prefix. *note Installation Names::
+ for more details, including other options available for fine-tuning
+ the installation locations.
+
+`--no-create'
+`-n'
+ Run the configure checks, but stop before creating any output
+ files.
+
+`configure' also accepts some other, not widely useful, options. Run
+`configure --help' for more details.
--- /dev/null
+DISTCHECK_CONFIGURE_FLAGS=--enable-gtk-doc
+
+SUBDIRS = \
+ gst \
+ common \
+ examples \
+ pkgconfig \
+ tests
+
+DIST_SUBDIRS = $(SUBDIRS)
+
+EXTRA_DIST = \
+ ChangeLog autogen.sh depcomp \
+ AUTHORS COPYING NEWS README RELEASE REQUIREMENTS \
+ gst-rtsp-server.spec docs/design/gst-rtp-server-design \
+ gst-rtsp-server.doap
+
+ACLOCAL_AMFLAGS = -I m4 -I common/m4
+
+DISTCLEANFILES = _stdint.h gst-rtsp-server.spec
+
+include $(top_srcdir)/common/release.mak
+include $(top_srcdir)/common/po.mak
+
+include $(top_srcdir)/common/coverage/lcov.mak
+
+check-valgrind:
+ cd tests/check && make check-valgrind
+
+if HAVE_CHECK
+check-torture:
+ cd tests/check && make torture
+else
+check-torture:
+ true
+endif
+
+# cruft: plugins that have been merged or moved or renamed
+CRUFT_FILES = \
+ $(top_builddir)/common/shave \
+ $(top_builddir)/common/shave-libtool \
+ $(top_builddir)/common/m4/codeset.m4 \
+ $(top_builddir)/common/m4/gettext.m4 \
+ $(top_builddir)/common/m4/glibc2.m4 \
+ $(top_builddir)/common/m4/glibc21.m4 \
+ $(top_builddir)/common/m4/iconv.m4 \
+ $(top_builddir)/common/m4/intdiv0.m4 \
+ $(top_builddir)/common/m4/intl.m4 \
+ $(top_builddir)/common/m4/intldir.m4 \
+ $(top_builddir)/common/m4/intlmacosx.m4 \
+ $(top_builddir)/common/m4/intmax.m4 \
+ $(top_builddir)/common/m4/inttypes-pri.m4 \
+ $(top_builddir)/common/m4/inttypes_h.m4 \
+ $(top_builddir)/common/m4/lcmessage.m4 \
+ $(top_builddir)/common/m4/lib-ld.m4 \
+ $(top_builddir)/common/m4/lib-link.m4 \
+ $(top_builddir)/common/m4/lib-prefix.m4 \
+ $(top_builddir)/common/m4/libtool.m4 \
+ $(top_builddir)/common/m4/lock.m4 \
+ $(top_builddir)/common/m4/longlong.m4 \
+ $(top_builddir)/common/m4/ltoptions.m4 \
+ $(top_builddir)/common/m4/ltsugar.m4 \
+ $(top_builddir)/common/m4/ltversion.m4 \
+ $(top_builddir)/common/m4/lt~obsolete.m4 \
+ $(top_builddir)/common/m4/nls.m4 \
+ $(top_builddir)/common/m4/po.m4 \
+ $(top_builddir)/common/m4/printf-posix.m4 \
+ $(top_builddir)/common/m4/progtest.m4 \
+ $(top_builddir)/common/m4/size_max.m4 \
+ $(top_builddir)/common/m4/stdint_h.m4 \
+ $(top_builddir)/common/m4/uintmax_t.m4 \
+ $(top_builddir)/common/m4/visibility.m4 \
+ $(top_builddir)/common/m4/wchar_t.m4 \
+ $(top_builddir)/common/m4/wint_t.m4 \
+ $(top_builddir)/common/m4/xsize.m4
+
+include $(top_srcdir)/common/cruft.mak
+
+all-local: check-cruft
--- /dev/null
+This is GStreamer RTSP Server 1.4.5
+
--- /dev/null
+gst-rtsp-server is a library ion top of GStreamer for building an RTSP server
+
+There are some examples in the examples/ directory and more comprehensive
+documentation in docs/README.
--- /dev/null
+
+Release notes for GStreamer RTSP Server Library 1.4.5
+
+The GStreamer team is pleased to announce a bugfix release of the stable
+1.4 release series. The 1.4 release series is adding new features on top
+of the 1.2 series and is part of the API and ABI-stable 1.x release
+series of the GStreamer multimedia framework that contains new features.
+The 1.4.x bugfix releases only contain important bugfixes compared to 1.4.0.
+
+
+Binaries for Android, iOS, Mac OS X and Windows are provided by the
+GStreamer project for this release.
+
+
+The 1.x series is a stable series targeted at end users. It is not API
+or ABI compatible with the 0.10.x series. It can, however, be installed
+in parallel with the 0.10.x series and will not affect an existing
+0.10.x installation.
+
+
+The stable 1.4.x release series is API and ABI compatible with 1.0.x and
+any other 1.x release series in the future. Compared to 1.0.x it contains
+some new features and more intrusive changes that were considered too
+risky as a bugfix.
+
+
+
+
+Bugs fixed in this release
+
+ * 739481 : rtsp-stream: leaks srtp decoder when leaving rtpbin
+
+==== Download ====
+
+You can find source releases of gst-rtsp-server in the download
+directory: http://gstreamer.freedesktop.org/src/gst-rtsp-server/
+
+The git repository and details how to clone it can be found at
+http://cgit.freedesktop.org/gstreamer/gst-rtsp-server/
+
+==== Homepage ====
+
+The project's website is http://gstreamer.freedesktop.org/
+
+==== Support and Bugs ====
+
+We use GNOME's bugzilla for bug reports and feature requests:
+http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer
+
+Please submit patches via bugzilla as well.
+
+For help and support, please subscribe to and send questions to the
+gstreamer-devel mailing list (see below for details).
+
+There is also a #gstreamer IRC channel on the Freenode IRC network.
+
+==== Developers ====
+
+GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned
+from there (see link above).
+
+Interested developers of the core library, plugins, and applications should
+subscribe to the gstreamer-devel list.
+
+
+Contributors to this release
+
+ * Aleix Conchillo Flaqué
+
\ No newline at end of file
--- /dev/null
+You need to have GStreamer. You can use an installed version of
+GStreamer or from its build dir.
+
--- /dev/null
+
+ - use a config file to configure the server
+ - error recovery
--- /dev/null
+#!/bin/sh
+# Run this to generate all the initial makefiles, etc.
+
+
+test -n "$srcdir" || srcdir=`dirname "$0"`
+test -n "$srcdir" || srcdir=.
+
+olddir=`pwd`
+cd "$srcdir"
+
+DIE=0
+package=gst-rtsp
+srcfile=gst/rtsp-server/rtsp-server.c
+
+# Make sure we have common
+if test ! -f common/gst-autogen.sh;
+then
+ echo "+ Setting up common submodule"
+ git submodule init
+fi
+git submodule update
+
+# source helper functions
+if test ! -f common/gst-autogen.sh;
+then
+ echo There is something wrong with your source tree.
+ echo You are missing common/gst-autogen.sh
+ exit 1
+fi
+. common/gst-autogen.sh
+
+# install pre-commit hook for doing clean commits
+if test ! \( -x .git/hooks/pre-commit -a -L .git/hooks/pre-commit \);
+then
+ rm -f .git/hooks/pre-commit
+ ln -s ../../common/hooks/pre-commit.hook .git/hooks/pre-commit
+fi
+
+NOCONFIGURE=1
+CONFIGURE_DEF_OPT='--enable-maintainer-mode --enable-gtk-doc'
+
+autogen_options $@
+
+printf "+ check for build tools"
+if test ! -z "$NOCHECK"; then echo ": skipped version checks"; else echo; fi
+version_check "autoconf" "$AUTOCONF autoconf autoconf270 autoconf269 autoconf268 " \
+ "ftp://ftp.gnu.org/pub/gnu/autoconf/" 2 68 || DIE=1
+version_check "automake" "$AUTOMAKE automake automake-1.11" \
+ "ftp://ftp.gnu.org/pub/gnu/automake/" 1 11 || DIE=1
+#version_check "autopoint" "autopoint" \
+# "ftp://ftp.gnu.org/pub/gnu/gettext/" 0 17 || DIE=1
+version_check "libtoolize" "$LIBTOOLIZE libtoolize glibtoolize" \
+ "ftp://ftp.gnu.org/pub/gnu/libtool/" 2 2 6 || DIE=1
+version_check "pkg-config" "" \
+ "http://www.freedesktop.org/software/pkgconfig" 0 8 0 || DIE=1
+
+die_check $DIE
+
+aclocal_check || DIE=1
+autoheader_check || DIE=1
+
+die_check $DIE
+
+# if no arguments specified then this will be printed
+if test -z "$*" && test -z "$NOCONFIGURE"; then
+ echo "+ checking for autogen.sh options"
+ echo " This autogen script will automatically run ./configure as:"
+ echo " ./configure $CONFIGURE_DEF_OPT"
+ echo " To pass any additional options, please specify them on the $0"
+ echo " command line."
+fi
+
+toplevel_check $srcfile
+
+# autopoint
+#if test -d po ; then
+# tool_run "$autopoint" "--force"
+#fi
+
+# aclocal
+# if test -f acinclude.m4; then rm acinclude.m4; fi
+
+tool_run "$libtoolize" "--copy --force"
+tool_run "$aclocal" "-I m4 -I common/m4 $ACLOCAL_FLAGS"
+tool_run "$autoheader"
+
+# touch the stamp-h.in build stamp so we don't re-run autoheader in maintainer mode
+echo timestamp > stamp-h.in 2> /dev/null
+
+tool_run "$autoconf"
+debug "automake: $automake"
+tool_run "$automake" "--add-missing --copy"
+
+test -n "$NOCONFIGURE" && {
+ echo "+ skipping configure stage for package $package, as requested."
+ echo "+ autogen.sh done."
+ exit 0
+}
+
+cd "$olddir"
+
+echo "+ running configure ... "
+test ! -z "$CONFIGURE_DEF_OPT" && echo " default flags: $CONFIGURE_DEF_OPT"
+test ! -z "$CONFIGURE_EXT_OPT" && echo " external flags: $CONFIGURE_EXT_OPT"
+echo
+
+echo "$srcdir/configure" $CONFIGURE_DEF_OPT $CONFIGURE_EXT_OPT
+"$srcdir/configure" $CONFIGURE_DEF_OPT $CONFIGURE_EXT_OPT || {
+ echo " configure failed"
+ exit 1
+}
+
+echo "Now type 'make' to compile $package."
--- /dev/null
+2008-12-17 Edward Hervey <bilboed@gmail.com>
+
+ * gst.supp:
+ And yet another variation of the GstAudioFilter leak.
+
+2008-12-15 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Patch by: Roland Illig <roland dot illig at gmx dot de>
+
+ * m4/gst-parser.m4:
+ Fix AG_GST_BISON_CHECK to handle version numbers with more than
+ two components (i.e. 2.4.1). Fixes bug #564507.
+
+2008-12-14 Edward Hervey <bilboed@gmail.com>
+
+ * gst.supp:
+ And yet another variant of the GstAudioFilter leak.
+
+2008-12-13 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst.supp:
+ Added variants of leaks of dynamic pad templates created in
+ GstAudioFilter.
+ Add conditional jump triggered by getaddrinfo (maybe glibc-2.9).
+
+2008-12-12 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst.supp:
+ Fix leak in GIO called by gnomevfs. Nothing we can do about this.
+
+2008-12-12 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst.supp:
+ Added another suppression for dynamic pad templates, in this case
+ GstAudioFilter.
+ Added suppression for PangoLanguage which can never be freed
+ according to the Pango API.
+
+2008-12-12 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst.supp:
+ A whole bunch of suppressions detected on latest gentoo ~amd64.
+ Make some existing suppressions more generic (for subtle dependecy
+ code changes).
+ Added suppressions for glibc-2.9.
+ Added suppressions for new variants of ALSA leaks.
+ Added suppressions for a series of leaks in plugins registrations due
+ to some pad templates' caps calculated at runtime.
+ Added suppressions for variants of some leaks in pango/fontconfig.
+ Added suppressions for leak in gstffmpegcsp.c (nothing we can do
+ about it, but will only exist once).
+
+2008-12-04 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * m4/gst-plugin-docs.m4:
+ Remove the check if $have_gtk_doc equals yes as it's not defined
+ and $enable_gtk_doc should be good enough.
+ Also this restores the build of the plugin documentation.
+
+2008-12-01 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst.supp:
+ Add suppression variant for Ubuntu Hardy x86/64bit.
+
+2008-12-01 Stefan Kost <ensonic@users.sf.net>
+
+ * gtk-doc-plugins.mak:
+ * gtk-doc.mak:
+ Simplily uninstall rule. Its closer to upstream and fixes #150331.
+
+2008-11-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * m4/glib-gettext.m4:
+ Update glib-gettext.m4 from latest stable GLib release.
+
+2008-11-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Patch by: Cygwin Ports maintainer
+ <yselkowitz at users dot sourceforge dot net>
+
+ * gettext.patch:
+ Update the gettext patch for use with gettext 0.17 which is
+ required to build with libtool 2.2 because of conflicts.
+ First part of bug #556091.
+
+2008-11-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * m4/gtk-doc.m4:
+ * m4/pkg.m4:
+ Update gtk-doc and pkg-config m4 macros from their latest releases.
+
+2008-11-20 Michael Smith <msmith@songbirdnest.com>
+
+ * m4/as-objc.m4:
+ Fix objective C test macro when none of the compilers are found at all.
+
+2008-10-30 Stefan Kost <ensonic@users.sf.net>
+
+ * gtk-doc.mak:
+ Also cp the entities here to all xinlcude based docs (workaround for
+ not being able to set up a search path).
+
+2008-10-17 Jan Schmidt <jan.schmidt@sun.com>
+
+ * gtk-doc.mak:
+ Don't clobber the real registry cache file when
+ building docs.
+
+2008-10-07 Jan Schmidt - Sun Microsystems <jan.schmidt@sun.com>
+
+ * m4/gst-error.m4:
+ Also disable the bogus "loop not entered at top" warnings appearing on Sparc Forte builds.
+
+2008-10-06 Stefan Kost <ensonic@users.sf.net>
+
+ * gtk-doc.mak:
+ Apply the same fix as below to gtk-doc.mak. Somehow did not end up in
+ CVS.
+
+2008-09-05 David Schleef <ds@schleef.org>
+
+ * gtk-doc-plugins.mak: Fix the check for gtkdoc-rebase: don't
+ pass the 'which' error back to make. This fix is more specific
+ than what is in upstream.
+
+2008-09-05 David Schleef <ds@schleef.org>
+
+ * gtk-doc.mak: Fix the check for gtkdoc-rebase: don't pass the
+ 'which' error back to make. This fix is more specific than
+ what is in upstream.
+
+2008-09-04 Stefan Kost <ensonic@users.sf.net>
+
+ * gtk-doc-plugins.mak:
+ * gtk-doc.mak:
+ Get closer to upstream makefiles. Don't install index.sgml twice. Call
+ gtkdoc-rebase (if exists).
+
+2008-08-21 Stefan Kost <ensonic@users.sf.net>
+
+ * gtk-doc-plugins.mak:
+ Revert $(top_builddir) -> $(builddir) change of rev. 1.39 as there is
+ no variable called builddir.
+
+2008-07-31 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst.supp:
+ Add suppressions for Ubunty Hardy x86/64bit, similar to earlier
+ versions and 32bit variant.
+
+2008-07-31 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * m4/gst-feature.m4:
+ Remove GST_DISABLE_(ENUMTYPES|INDEX|URI).
+
+2008-07-21 Tim-Philipp Müller <tim.muller at collabora co uk>
+
+ * m4/gst-error.m4::
+ When checking for GST_ERROR_CXXFLAGS, check each compiler flag
+ individually, not all together.
+
+2008-07-20 Tim-Philipp Müller <tim.muller at collabora co uk>
+
+ * m4/gst-parser.m4::
+ Fix bison version number detection for older --version
+ output format (as bison 1.28 on OSX 10.4 outputs).
+ Fixes #543853.
+
+2008-07-12 Stefan Kost <ensonic@users.sf.net>
+
+ * plugins.xsl:
+ Split refsect2 also here to make "Element Pads" subtitle visible.
+
+2008-07-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * m4/gst-error.m4:
+ Add compiler flags to warn if declarations after statements or
+ variable length arrays are used. These are C99/GCC extensions and
+ are not supported by some compilers we want to support.
+
+2008-07-02 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gtk-doc-plugins.mak:
+ Only clean doc maintainer stamps in maintainer-clean. Fixes #539977.
+
+2008-06-20 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gstdoc-scangobj:
+ Always use format strings for printf-like functions, even if they just
+ print a string. Fixes bug #536981.
+
+2008-06-20 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gtk-doc-plugins.mak:
+ * gtk-doc.mak:
+ Include CFLAGS and LDFLAGS in GTKDOC_CFLAGS and GTKDOC_LDFLAGS,
+ otherwise the values passed to configure are ignored.
+ Fixes bug #536978.
+
+2008-06-05 Tim-Philipp Müller <tim.muller at collabora co uk>
+
+ * m4/gst-error.m4:
+ Add -fno-strict-aliasing when compiling with -Werror, to work around
+ warnings caused by G_LOCK with recent GLib versions (2.16.x) (#316221).
+
+2008-06-05 Jan Schmidt <jan.schmidt@sun.com>
+
+ * gtk-doc.mak:
+ Don't copy html/*.png files unless they don't already exist
+ in the destdir. Fixes distcheck failure caused by permissions
+ problems trying to copy a file into the destdir when it already
+ exists.
+
+2008-05-28 Stefan Kost <ensonic@users.sf.net>
+
+ * plugins.xsl:
+ The class was not shown in plugin docs. Fix typo in changelog below.
+
+2008-05-22 Jan Schmidt <jan.schmidt@sun.com>
+
+ * gstdoc-scangobj:
+ Emit warnings if one of the GTypes we're expecting is 0
+ when scanning.
+
+2008-05-21 Felipe Contreras <felipe.contreras@gmail.com>
+
+ * gtk-doc-plugins.mak:
+ * gtk-doc.mak:
+ Fix installing png images when gtk-doc is disabled.
+
+2008-05-21 Felipe Contreras <felipe.contreras@gmail.com>
+
+ * gtk-doc-plugins.mak:
+ * gtk-doc.mak:
+ Fix make clean when gtk-doc is disabled and other cleanups.
+
+2008-05-17 Jan Schmidt <jan.schmidt@sun.com>
+
+ * gtk-doc-plugins.mak:
+ Be more quiet when the files don't yet exist.
+
+2008-05-16 Jan Schmidt <jan.schmidt@sun.com>
+
+ * gstdoc-scangobj:
+ Add a mechanism for adding 'implicitly created' GTypes into the
+ scan, allowing for documenting plugin-private base classes that
+ provide signals or properties for public elements.
+
+ * gtk-doc-plugins.mak:
+ Use $(builddir) instead of $(top_builddir) in a few places - there's
+ no need to hard code 'docs/plugins' as the only useable path.
+
+2008-05-14 Peter Kjellerstedt <pkj@axis.com>
+
+ * m4/gst-feature.m4:
+ Report plug-ins without external dependencies that will not be built
+ even when the name of the plug-in is a substring of another plug-in,
+ e.g., goom vs. goom2k1.
+
+2008-05-14 Tim-Philipp Müller <tim.muller at collabora co uk>
+
+ * gst.supp:
+ Add suppression for glibc bug on gutsy/x86-64
+
+2008-05-12 Stefan Kost <ensonic@users.sf.net>
+
+ * plugins.xsl:
+ Improve the layout of the caps, but splitting them on ";".
+
+2008-05-09 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Patch by: Brian Cameron <brian dot cameron at sun dot com>
+
+ * m4/gst-default.m4:
+ Don't set the default audio sink to the default visualizer.
+ Fixes bug #532295.
+
+2008-05-07 Tim-Philipp Müller <tim.muller at collabora co uk>
+
+ * check.mak: (help):
+ Document GST_CHECKS environment variable in checks 'make help'.
+
+2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Patch by: Marc-Andre Lureau <marcandre dot lureau at gmail dot com>
+
+ * scangobj-merge.py:
+ Don't depend on Twisted just for the OrderedDict but implement our
+ own ordered dictionary class. Fixes bug #531577.
+
+2008-04-23 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst.supp:
+ Re-arrange latest suppressions.
+ Add all known suppressions for ubuntu hardy. Same as for older
+ ubuntus, but with different codepaths.
+
+2008-04-22 Edward Hervey <bilboed@gmail.com>
+
+ * gst.supp: Make tls leak suppression a bit more generic.
+
+2008-04-22 Edward Hervey <bilboed@gmail.com>
+
+ * gst.supp: Fix ommission in latest commit.
+ Make tls leak suppression more generic in order to cover more
+ distributions (and hopefully also future distributions).
+
+2008-04-22 Edward Hervey <bilboed@gmail.com>
+
+ * gst.supp: Add suppressions for Hardy.
+ They're just the newer versions of similar suppressions we had
+ for the previous versions of ubuntu.
+
+2008-04-15 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * Makefile.am:
+ * m4/Makefile.am:
+ Dist all files in common. Fixes bug #527984.
+
+2008-04-14 Tim-Philipp Müller <tim at centricular dot net>
+
+ * m4/gst-function.m4:
+ Rename AC_CACHE_VAL cache-ids to contain '_cv_' in order to make
+ autoconf-2.62 complain less.
+
+2008-04-13 Tim-Philipp Müller <tim at centricular dot net>
+
+ * m4/gst-args.m4:
+ * m4/gst-valgrind.m4:
+ Bump valgrind requirement to 3.0 (which was released in August 2005).
+ Fixes #489269. Also, check for version >=REQ and not >REQ.
+
+2008-04-09 Tim-Philipp Müller <tim at centricular dot net>
+
+ * m4/gst-default.m4:
+ Add --with-default-{audiosink|audiosrc|videosink|videosrc|visualizer}
+ configure switches (#519417).
+
+2008-04-03 Tim-Philipp Müller <tim at centricular dot net>
+
+ * m4/gst-args.m4:
+ Add --disable-foo switch for dependency-less plugins (#525586).
+
+2008-04-01 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * m4/gst-parser.m4:
+ Unconditionally require flex 2.5.31 and bison 1.875.
+
+2008-03-23 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * m4/gst-arch.m4:
+ amd64/x86_64 allows unaligned memory access too.
+
+2008-03-21 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * m4/gst-dowhile.m4:
+ Add macro that checks if the compiler supports do {} while (0)
+ macros and define HAVE_DOWHILE_MACROS if it does. This is
+ needed by glib/gmacros.h to use something else than
+ if (1) else for G_STMT_START/END when compling C++, which
+ causes compiler warnings because of ambigious else with g++ 4.3.
+
+2008-03-21 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * m4/gst-plugin-docs.m4:
+ * mangle-tmpl.py:
+ Don't depend on PyXML and use only XML modules that are shipped
+ with python. Fixes bug #519635.
+
+2008-03-07 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * m4/gtk-doc.m4: (GTK_DOC_CHECK):
+ The previous commit to this file by Stefan Kost mentionned checking for
+ SED, but NOT checking for gtkdoc-check (wth is that doing there ??).
+ Therefore, removing the check for gtkdoc-check
+
+2008-03-03 David Schleef <ds@schleef.org>
+
+ * m4/ax_create_stdint_h.m4: Oops, checked in the wrong copy of
+ this file. (Update from upstream)
+
+2008-03-03 David Schleef <ds@schleef.org>
+
+ * m4/ax_create_stdint_h.m4: Update from upstream. Fixes a bug
+ compiling with MSVC.
+
+2008-03-03 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * m4/pkg.m4:
+ Allow override of pkg-config results, as proposed by configure --help.
+ This is in fact just a backport from upstream pkg.m4.
+ Fixes #518892
+
+2008-03-03 Peter Kjellerstedt <pkj@axis.com>
+
+ * ChangeLog:
+ Changelog surgery of my previous commit to add bugzilla reference.
+ * m4/gst-args.m4:
+ Add AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to make it easier
+ to include and exclude plug-ins without external references, i.e.,
+ plug-ins listed in GST_PLUGINS_SELECTED. (#498222)
+
+2008-03-03 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst.supp:
+ Add another glibc suppression.
+
+2008-02-29 Peter Kjellerstedt <pkj@axis.com>
+
+ * m4/gst-feature.m4:
+ Make the comment before defines generated via AG_GST_CHECK_FEATURE
+ look nicer. (#498222)
+
+2008-02-26 Jan Schmidt <jan.schmidt@sun.com>
+
+ * m4/Makefile.am:
+ * m4/as-gcc-inline-assembly.m4:
+ Add Dave Schleef's GCC inline assembly detection macro
+ for using in gst-plugins-good in the goom 2k4 plugin.
+
+2008-02-25 Andy Wingo <wingo@pobox.com>
+
+ * gst-autogen.sh: Instead of only passing certain arguments to
+ configure, pass anything that we didn't handle. Much friendlier.
+ Fixes #34412.
+
+2008-02-23 Jan Schmidt <Jan.Schmidt@sun.com>
+
+ * m4/gst-error.m4:
+ Store the detected compiler flags into ERROR_CFLAGS rather than
+ ERROR_CXXFLAGS, and use the macro that checks the C compiler, not
+ the C++ one.
+
+2008-02-23 Tim-Philipp Müller <tim at centricular dot net>
+
+ * m4/gst-error.m4:
+ Reflow checks for additional warning flags so they're not
+ nested, which fixes the result reporting in the configure
+ output.
+
+2008-02-22 Tim-Philipp Müller <tim at centricular dot net>
+
+ * m4/as-compiler-flag.m4:
+ Add AS_CXX_COMPILER_FLAG
+
+ * m4/gst-error.m4:
+ Add AG_GST_SET_ERROR_CXXFLAGS (Forte bits need testing)
+
+2008-02-22 Tim-Philipp Müller <tim at centricular dot net>
+
+ * gtk-doc-plugins.mak:
+ Add 'check-inspected-versions' target; this helps identify
+ files that should have been removed or where the version
+ number should (ideally) be updated before a release
+ (which doesn't happen automatically if the releaser doesn't
+ build that plugin locally). Not adding at a distcheck hook
+ yet though, because it's not really that important and would
+ probably also be a problem on buildbots.
+
+2008-02-22 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst.supp:
+ Add even more glibc 2.7 suppressions.
+
+2008-02-22 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst.supp:
+ Add another suppression for GLib caching some values after
+ the first call.
+
+2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Patch by:
+ Tim Mooney <mooney at dogbert dot cc dot ndsu dot nodak dot edu>
+
+ * m4/gst-error.m4:
+ Use no%E_MACRO_REDEFINED on Solaris to prevent compiler warnings.
+ Fixes bug #515905.
+
+2008-02-11 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst.supp:
+ Add a few more glibc 2.7 suppressions to make the avisubtitle unit
+ test valgrind clean. Fixes bug #515703.
+
+2008-02-08 Stefan Kost <ensonic@users.sf.net>
+
+ * ChangeLog:
+ Changelog surgery for last commit.
+
+2008-02-08 Stefan Kost <ensonic@users.sf.net>
+
+ * m4/gtk-doc.m4:
+ Conditionally check for SED. Also sync a bit with upstream macro.
+
+2008-02-08 Stefan Kost <ensonic@users.sf.net>
+
+ * gtk-doc-plugins.mak:
+ * gtk-doc.mak:
+ Use '$(SED)' instead of 'sed'. Don't use -i for in-place as its gnu
+ only, move to a temp file instead.
+
+2008-02-06 Stefan Kost <ensonic@users.sf.net>
+
+ * gtk-doc-plugins.mak:
+ * gtk-doc.mak:
+ As our docs are versioned, we need to patch the index.sgml file to have
+ correct paths there, unless we also want to fork gtk-doc's xsl (which
+ we don't). This hopefully fixes xrefs between modules.
+
+2008-02-02 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * m4/gst-feature.m4:
+ Use printf instead of echo as "echo -e" isn't POSIX and doesn't work
+ with strict POSIX shells like tcsh or dash and also not every platform
+ has a /bin/echo that supports it.
+
+2008-01-24 Stefan Kost <ensonic@users.sf.net>
+
+ * ChangeLog:
+ ChangeLog surgery.
+
+ * gstdoc-scangobj:
+ Sync the object scanner with gtk-doc fixes. Update args and hierarchy
+ files.
+
+2008-01-20 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * check.mak:
+ * coverage/lcov.mak:
+ * gtk-doc-plugins.mak:
+ * release.mak:
+ Use $(MAKE) instead of make to fix the build if GNU make is called
+ something else on the system.
+
+ * m4/as-docbook.m4:
+ Fix path for docbook.xsl if we have no /etc/xml/catalog and add a
+ docbook-xsl search path for FreeBSD.
+
+2008-01-18 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst.supp:
+ Add a suppression for a glibc bug:
+ http://valgrind.org/docs/manual/faq.html#faq.exit_errors>
+
+2008-01-18 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst.supp:
+ Add some more glibc 2.7 suppressions and make the GLib suppressions
+ for the home/tmp/etc directory caching a bit more generic.
+
+2008-01-18 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst.supp:
+ Add some glibc 2.7 supressions as found on Debian/unstable.
+
+2008-01-14 Jan Schmidt <jan.schmidt@sun.com>
+
+ * download-translations:
+ Apparently I have problems with leaving things commented out when
+ I edit shell scripts.
+
+2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+
+ * download-translations:
+ Remove bash-isms
+
+2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+
+ * check-exports:
+ Restore the cleanup rm of our tmp file which I didn't mean to leave
+ commented out.
+
+2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+
+ * check-exports:
+ Fixes to make check-export work on both Solaris and Linux
+
+ * m4/gst-error.m4:
+ Disable extra warning category (argument mismatch) as an error
+ on Forte, as it prevents the libcheck fail_if macros from compiling.
+
+ * win32.mak:
+ Substitute the GStreamer version so things will keep working in 0.11
+
+2008-01-11 Tim-Philipp Müller <tim at centricular dot net>
+
+ Patch by: Peter Kjellerstedt <pkj axis com>
+
+ * m4/gst-glib2.m4:
+ * m4/gst-libxml2.m4:
+ Improve/fix output from configure if either glib-2.0 or
+ libxml2 are not installed (#498222).
+
+2008-01-09 Stefan Kost <ensonic@users.sf.net>
+
+ * coverage/lcov.mak:
+ Update coverage make-rules: use them conditionaly, use libtool mode
+ and use lcov to cleanup.
+
+2007-12-18 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * glib-gen.mak:
+ Also use #include "header" instead of #include <header> for the
+ headers that were used to generate the source files for the same
+ reason as below.
+
+ Remove whitespace before #include.
+
+2007-12-18 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * glib-gen.mak:
+ Use #include "header" instead of #include <header> for the generated
+ enum C files as the file will always be in the same directory and
+ some compilers seem to be a bit strict about that unless . is added
+ to the include path.
+
+ Include all headers that were used to generate the source files in
+ the C file as they're used there.
+
+2007-12-17 Tim-Philipp Müller <tim at centricular dot net>
+
+ * win32.mak: (win32), (win32defs), (win32crlf):
+ Make check for CR LF in Visual C++ 6.0 project files
+ work, based on patch by David Schleef (#496722, #393626).
+
+2007-12-17 Tim-Philipp Müller <tim at centricular dot net>
+
+ * Makefile.am:
+ Don't forget to dist the new win32.mak.
+
+2007-12-17 Tim-Philipp Müller <tim at centricular dot net>
+
+ * win32.mak: (win32), (win32defs):
+ Move common win32 Makefile foo into this new file.
+
+2007-12-15 Stefan Kost <ensonic@users.sf.net>
+
+ * gtk-doc-plugins.mak:
+ * gtk-doc.mak:
+ We should have never forked this that much :/.
+
+2007-12-13 Tim-Philipp Müller <tim at centricular dot net>
+
+ * check-exports:
+ Fix build on the ppc64 build bot.
+
+2007-12-13 Tim-Philipp Müller <tim at centricular dot net>
+
+ * check-exports:
+ Suppress more unintentional exports (too much hassle to rename them,
+ since the win32 project files would need changing too).
+
+2007-12-12 Tim-Philipp Müller <tim at centricular dot net>
+
+ * Makefile.am:
+ check-exports should be disted.
+
+2007-12-12 Tim-Philipp Müller <tim at centricular dot net>
+
+ * check-exports:
+ Add quick'n'dirty script to check the exported symbols of a library
+ against the symbols in the corresponding .def file (#493983). Based
+ on script by Ole André Vadla Ravnås.
+
+2007-11-06 Jan Schmidt <jan.schmidt@sun.com>
+
+ * gtk-doc-plugins.mak:
+ Fix distcheck by making sure the types files are treated like the
+ other gtkdoc-scangobj generated files.
+
+2007-09-21 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * m4/gst-args.m4:
+ Let the AG_GST_ARG_ENABLE_EXPERIMENTAL macro default to disable
+ building of experimental plugins. Nobody uses it yet and the
+ --enable--experimental stuff from gst-plugins-good defaults to
+ disable too.
+
+2007-09-06 Tim-Philipp Müller <tim at centricular dot net>
+
+ * gtk-doc-plugins.mak:
+ Just use the normal 'check' target and avoid a circular
+ dependency.
+
+2007-09-06 Tim-Philipp Müller <tim at centricular dot net>
+
+ * gtk-doc-plugins.mak:
+ Add rule to error out if .hierarchy file contains tabs.
+
+2007-08-20 Tim-Philipp Müller <tim at centricular dot net>
+
+ * download-translations:
+ * po.mak:
+ If there are new languages, they need to be added to po/LINGUAS.
+
+2007-08-20 Tim-Philipp Müller <tim at centricular dot net>
+
+ * download-translations:
+ * po.mak:
+ Fix up 'download-po' a bit, so that we find new translations
+ for languages that aren't in our po/LINGUAS file yet too.
+
+2007-07-16 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * gst.supp:
+ Add a suppression for GLib caching the tmp dir seen on an
+ Ubuntu Feisty system.
+
+2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * m4/gst-feature.m4:
+ If we want to use 'echo -e', call /bin/echo instead of the shell's
+ since -e is a bash extension, and our /bin/sh might not be being
+ provided by bash.
+
+2007-07-01 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * po.mak:
+ Translation project has moved. Also, no idea how this used to
+ work given that we weren't downloading a .po file.
+
+2007-06-25 Stefan Kost <ensonic@users.sf.net>
+
+ * gst-xmlinspect.py:
+ * plugins.xsl:
+ Also extract element caps for plugin-docs. Fixes parts of #117692.
+
+2007-06-21 Tim-Philipp Müller <tim at centricular dot net>
+
+ Patch by: Andreas Schwab
+
+ * m4/gst-feature.m4:
+ Fix quoting (#449493).
+
+2007-06-10 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * m4/gst-parser.m4:
+ Only generate the parser if bison >= 1.875 _and_ flex >= 2.5.31 is
+ installed and use pre-generated sources otherwise. Fixes bug #444820.
+
+2007-05-11 Michael Smith <msmith@fluendo.com>
+
+ * gst.supp:
+ Suppression variant for our good friend the TLS leak, this time for
+ Ubuntu Feisty/x86.
+
+2007-05-09 Tim-Philipp Müller <tim at centricular dot net>
+
+ * gtk-doc-plugins.mak:
+ Fix make distcheck again; change some spaces to tabs in makefile.
+
+2007-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gtk-doc-plugins.mak (-module):
+ Error out when the html build step gives warnings, so they get
+ fixed properly.
+
+2007-04-23 Stefan Kost <ensonic@users.sf.net>
+
+ * m4/gst-feature.m4:
+ Add macro AG_GST_PARSE_SUBSYSTEM_DISABLES that checks the defines in
+ the configuration header and AC_DEFINES the setings.
+
+2007-04-19 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Patch by: Vincent Torri <vtorri at univ-evry dot fr>
+
+ * m4/gst-parser.m4:
+ Put the AC_MSG_RESULT output in brackets to get it properly written to
+ the terminal.
+
+2007-04-18 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * m4/gst-parser.m4:
+ Check for flex >= 2.5.31 and set GENERATE_PARSER if we have at least
+ that version. Otherwise use pre-generated parser sources as we can't
+ raise the required flex version. HAVE_MT_SAVE_FLEX is obsolete now
+ as we use a new enough flex version anyway. First part of #349180
+
+2007-04-10 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-check.m4:
+ Allow pre-setting the GST(PB)_TOOLS/PLUGINS_DIR variables to help
+ builds against older GStreamer.
+
+2007-03-25 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * m4/gst-parser.m4:
+ Fix the flex version check. It ignored the micro version before.
+
+2007-03-09 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * check.mak:
+ Use the same timeout when generating valgrind suppressions as
+ running the valgrind test.
+
+ * gst.supp:
+ Add some more suppressions and stuff.
+
+2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * check.mak:
+ Make sure GSlice is disabled when building suppressions too.
+
+ * gst.supp:
+ Add around *850* lines of suppressions for one-time initialisations
+ inside libasound and gconf/bonobo/ORBit. I feel so dirty.
+
+2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * gst.supp:
+ add a suppression for this GConf flup on the FC5 buildbot.
+
+2007-03-06 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * gst.supp:
+ Make the suppression a little more generic, to catch the FC5
+ backtrace too.
+
+2007-03-06 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * gst.supp:
+ Add a suppression for libcdio 0.76. It leaks an internal struct
+ when the CD-ROM device is not accessible.
+
+2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-arch.m4:
+ Move a line that was in the wrong macro
+
+2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst.m4:
+ Add
+ * m4/gst-arch.m4:
+ * m4/gst-args.m4:
+ * m4/gst-check.m4:
+ * m4/gst-debuginfo.m4:
+ * m4/gst-default.m4:
+ * m4/gst-doc.m4:
+ * m4/gst-error.m4:
+ * m4/gst-feature.m4:
+ * m4/gst-function.m4:
+ * m4/gst-gettext.m4:
+ * m4/gst-glib2.m4:
+ * m4/gst-libxml2.m4:
+ * m4/gst-parser.m4:
+ * m4/gst-plugin-docs.m4:
+ * m4/gst-plugindir.m4:
+ * m4/gst-valgrind.m4:
+ * m4/gst-x11.m4:
+ Convert all macros to use AG_GST style so we can properly warn
+ when they're missing if configure.ac calls AG_GST_INIT
+ Will require update in all GStreamer modules.
+
+2007-02-11 Stefan Kost <ensonic@users.sf.net>
+
+ * m4/gst-args.m4:
+ Remove 'enable' from configure switch description as this leads to
+ confusing lines like "disable enable builing ...".
+ * m4/gst-feature.m4:
+ Fix comment to sound less horrible.
+
+2007-02-07 Tim-Philipp Müller <tim at centricular dot net>
+
+ Patch by: Will Newton <will.newton gmail com>
+
+ * m4/gst-check.m4:
+ Use $PKG_CONFIG rather than pkg-config directly, the one in our path
+ might not be the one we want, like when cross-compiling. Also, other
+ macros such as PKG_CHECK_MODULES use $PKG_CONFIG, so we should
+ probably too just for consistency. Fixes #405288.
+
+2007-01-08 Tim-Philipp Müller <tim at centricular dot net>
+
+ * m4/gst-parser.m4:
+ Need to use double square brackets again so m4 doesn't remove them
+ (fixes #378931).
+
+ * m4/gst-args.m4:
+ Use double square brackets here as well, for the same reason.
+
+2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
+
+ * m4/gst-parser.m4:
+ Use 'sed' rather than 'tr' to strip trailing letters from version
+ numbers, since 'tr' might not be available and we know sed is
+ (#378931).
+
+2006-10-21 Tim-Philipp Müller <tim at centricular dot net>
+
+ * check.mak:
+ Increase default timeout under valgrind, 60 is just too short and
+ some tests take a bit longer these days and not everyone has a
+ beefy machine.
+
+2006-09-29 Michael Smith <msmith@fluendo.com>
+
+ * gst.supp:
+ More suppressions for edgy.
+
+2006-09-28 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * m4/gst-glib2.m4:
+ Use gmodule-no-export-2.0.pc instead of gmodule-2.0.pc - we neither
+ want nor need --export-dynamic (which ends up making us export a bunch
+ of unneeded symbols)
+
+2006-09-14 Tim-Philipp Müller <tim at centricular dot net>
+
+ * gst.supp:
+ Some suppressions for the more recent ld.so in ubuntu edgy.
+
+2006-08-23 Tim-Philipp Müller <tim at centricular dot net>
+
+ * gst.supp:
+ Shorten function trail so the suppression works on
+ my ubuntu dapper system with core cvs as well.
+
+2006-07-28 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * gst.supp:
+ Extra suppressions from my Ubuntu x86_64 machine
+
+2006-07-24 Tim-Philipp Müller <tim at centricular dot net>
+
+ Patch by: Frederic Peters <fpeters at entrouvert com>
+
+ * m4/gst-parser.m4:
+ Need to double square brackets in .m4 files. Should fix bison
+ version detection with version numbers like 1.23a (#348354).
+
+2006-07-24 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * check.mak:
+ Valgrind fails to find tests written in tests/check/ directly (rather
+ than a subdir) - because valgrind gets run with a filename that
+ doesn't contain a relative path, it goes searching /usr/bin instead.
+ Run with ./.... to make things work either way.
+
+ * gtk-doc-plugins.mak:
+ Add $(top_builddir)/src as a place to look for plugins
+ when building too, since that's where gst-template keeps things
+
+2006-07-23 Stefan Kost <ensonic@users.sf.net>
+
+ Patch by: Frederic Peters <fpeters@entrouvert.com>
+
+ * m4/gst-parser.m4:
+ Fix bison detection (#348354)
+
+2006-07-21 Stefan Kost <ensonic@users.sf.net>
+
+ * m4/gst-parser.m4:
+ check for bison and flex
+
+2006-07-13 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-plugin-docs.m4:
+ remove the configure argument for enabling plugin doc build;
+ having gtk-doc enabled and pyxml present is enough of a trigger
+
+2006-07-03 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * coverage/lcov.mak:
+ fix up rules to work with gst-python as well
+ run "make lcov" to test and generate the reports
+ run "make lcov-reset" to redo it after that
+
+2006-07-02 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * Makefile.am:
+ * check.mak:
+ add an inspect target that inspects every element feature,
+ so we can have that added for coverage
+ * coverage/lcov.mak:
+ add support for lcov
+
+2006-07-02 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-args.m4:
+ when building with gcov, reset CFLAGS and friends to O0
+
+2006-07-02 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-args.m4:
+ Find the gcov that matches the gcc version
+ Only allow gcov if we use gcc
+
+2006-07-02 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * Makefile.am:
+ * coverage/coverage-report-entry.pl:
+ * coverage/coverage-report.pl:
+ * coverage/coverage-report.xsl:
+ copy coverage reporting files from dbus
+
+2006-07-01 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-args.m4:
+ libtool strips gcov's -f flags, so libgcov does not get
+ linked in. Setting GCOV_LIBS with -lgcov fixes libtool's
+ stripping
+ also show what pkg-config-path we set
+
+2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+
+ Patch by: Peter Kjellerstedt <pkj at axis com>
+
+ * m4/gst-feature.m4:
+ Show list of plugins without external dependencies that
+ will not be built as well (#344136).
+
+2006-06-15 Tim-Philipp Müller <tim at centricular dot net>
+
+ * m4/gst-plugin-docs.m4:
+ add GST_PLUGIN_DOCS, which checks for everything needed
+ to build the plugin docs (namely gtk-doc and pyxml); also
+ adds a new --enable-plugin-docs configure switch; will
+ set ENABLE_PLUGIN_DOCS conditional for use in Makefile.am
+ files (see #344039).
+
+2006-06-11 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-check.m4:
+ add GST_PKG_CHECK_MODULES, which in the normal case of checking
+ for a dependency lib for a plug-in only needs two arguments
+ to do the right thing.
+ * m4/gst-feature.m4:
+ clean up output a little of feature checking; also deal with
+ non-plug-in feature checks
+ * m4/Makefile.am:
+ * m4/gst-gstreamer.m4:
+ remove this file; it's a useless check
+
+2006-06-06 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-arch.m4:
+ add PPC64 so we can have separate structure sizes for it
+
+2006-06-05 Edward Hervey <edward@fluendo.com>
+
+ * gtk-doc.mak:
+ Check for the proper .devhelp2 file to remove.
+
+2006-05-31 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gtk-doc.mak:
+ allow a magic variable to suppress errors from docbuilding
+
+2006-05-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+
+ * gtk-doc.mak:
+ error out if gtkdoc-mktmpl finds unused declarations
+
+2006-05-28 Edward Hervey <edward@fluendo.com>
+
+ * gst.supp:
+ Reverting previous commit. That's good to know, Edward, but why ?
+
+2006-05-28 Edward Hervey <edward@fluendo.com>
+
+ * gst.supp:
+ Added suppresion for memleak in g_option_context_parse on fc5-64
+
+2006-05-19 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-check.m4:
+ set GSTPB_PLUGINS_DIR just like GST_PLUGINS_DIR
+
+2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
+
+ * check.mak:
+ Fix 'make help' in check directories, it should be
+ 'valgrind.gen-suppressions' not 'valgrind-gen-suppressions'
+ (not changing target to match help string on purpose to keep
+ scripts etc. functional).
+
+2006-05-18 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ Patch by: Peter Kjellerstedt
+
+ * m4/gst-arch.m4:
+ add support for CRIS and CRISv32.
+
+2006-05-17 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * m4/gst-args.m4:
+ Fix the macros for command-line supplied package and origin names
+ so they don't end up being configure as "" (Fixes #341479)
+
+2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * gtk-doc.mak:
+ Add uninstall rule to remove .devhelp2 files.
+
+2006-05-09 Edward Hervey <edward@fluendo.com>
+
+ * gst.supp:
+ Add suppression for GSlice version of
+ g_type_init calloc leak
+
+2006-04-05 Michael Smith <msmith@fluendo.com>
+
+ * gst.supp:
+ Delete a bogus suppression for the registry code.
+ Generalise a suppression for a glib bug (see #337404)
+
+2006-04-04 Michael Smith <msmith@fluendo.com>
+
+ * gst.supp:
+ Add a leak suppression: the existing glibc-doesn't-free-TLS one
+ wasn't triggering here.
+
+2006-04-04 Michael Smith <msmith@fluendo.com>
+
+ * gst.supp:
+ Add some minimally-neccesary suppressions for my x86/dapper system.
+
+2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * plugins.xsl:
+ Do not display an origin link if origin does not start with http
+ See #323798
+
+2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-args.m4:
+ * m4/gst-feature.m4:
+ add more macros
+ * m4/gst-x11.m4:
+ X11-related checks
+
+2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/as-version.m4:
+ newer version
+ * m4/gst-args.m4:
+ * m4/gst-doc.m4:
+ update and add other macros to be shared across projects
+
+2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gst.supp:
+ add a suppression for g_parse_debug_string
+
+2006-03-23 Stefan Kost <ensonic@users.sf.net>
+
+ * gstdoc-scangobj:
+ sync fully with gtkdoc-0.15
+
+2006-03-23 Stefan Kost <ensonic@users.sf.net>
+
+ * gstdoc-scangobj:
+ * gtk-doc.mak:
+ sync a little with gtk-doc mainline
+
+2006-03-17 Wim Taymans <wim@fluendo.com>
+
+ * gst.supp:
+ add another clone suppression
+ change all glibc suppressions to match 2.3.*
+
+2006-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/check.m4:
+ fix test so it actually works when the normal check is used
+ over debian's/ubuntu's
+
+2006-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * check.mak:
+ Set G_SLICE=always-malloc when valgrinding tests
+ (closes #333272)
+
+2006-02-21 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * m4/gst-glib2.m4:
+ Fix debug output when the GLib version prerequisite is not found
+
+2006-02-13 Andy Wingo <wingo@pobox.com>
+
+ * m4/check.m4: Hack around Debian/Ubuntu's broken installation of
+ the PIC version of check as libcheck_pic.a. Should work with
+ cross-compilation too. Grr.
+
+2006-02-06 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-default.m4:
+ switch to auto* sinks for defaults
+
+2006-02-02 Wim Taymans <wim@fluendo.com>
+
+ * check.mak:
+ add a .valgrind.gen-suppressions target to aid in generating
+ suppressions
+ * gst.supp:
+ add more repressions from my debian glibc as of today
+
+2006-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gtk-doc-plugins.mak:
+ only add srcdir/gst if it exists
+
+2006-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * release.mak:
+ don't complain about disted enums in win32
+
+2006-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-check.m4:
+ AC_SUBST CFLAGS and LIBS
+ do a non-command because something is stripping out our AC_SUBST
+
+2006-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-args.m4:
+ * m4/gst-valgrind.m4:
+ properly give a "no" result manually when providing a
+ not-found action to fix configure output
+
+2006-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/pkg.m4:
+ update with a more recent version
+
+2006-01-07 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gettext.patch:
+ make Makefile depend on LINGUAS, so rebuilds work when adding
+ a language
+
+2006-01-03 Michael Smith <msmith@fluendo.com>
+
+ * check.mak:
+ Clarify error message from valgrind test runs.
+
+2005-12-16 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-arch.m4:
+ define HOST_CPU
+
+2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * check.mak:
+ add a valgrind-forever target for tests
+
+2005-11-28 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * check.mak:
+ when a "make test.check" run fails, make it rerun the test with
+ at least debug level 2
+
+2005-11-14 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/Makefile.am:
+ * m4/gst-check.m4:
+ fix check for base plugins
+ * m4/gst-default.m4:
+ add m4 to set default elements
+
+2005-10-18 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-check.m4:
+ check for tools correctly
+
+2005-10-18 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gtk-doc.mak:
+ only enable breaking on new API when make distcheck passes,
+ not before
+
+2005-10-18 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-check.m4:
+ Resurrect Julien's dead body and wipe his mind clean
+
+2005-10-18 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-check.m4:
+ Kill Julien
+
+2005-10-17 Julien MOUTTE <julien@moutte.net>
+
+ * m4/gst-check.m4: I know Thomas will kill me but this
+ ifelse statement seems incorrect as it is always setting
+ required to "yes". With this one it seems to work. Fixes
+ build of gst-plugins-base on my setup where gstreamer-check
+ is definitely not present/required.
+
+2005-10-18 Stefan Kost <ensonic@users.sf.net>
+
+ * gtk-doc.mak:
+ make build break on new api that has not been added to the
+ sections file
+
+2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-glib2.m4:
+ * m4/Makefile.am:
+ * m4/gst-check.m4:
+ add macro for easy checks for GStreamer libs
+
+2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-glib2.m4:
+ update, warn in error cases
+
+2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-error.m4:
+ add GST_SET_DEFAULT_LEVEL
+
+2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/Makefile.am:
+ * m4/gst-gettext.m4:
+ remove the AM_GNU_GETTEXT* calls, they need to be in configure.ac
+ * m4/gst-glib2.m4:
+ clean up and re-use in core soon
+ * m4/gst-plugindir.m4:
+ macro to set up PLUGINDIR and plugindir define/var
+
+2005-10-15 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/Makefile.am:
+ * m4/gst-gettext.m4:
+ add macro for setting up gettext
+
+2005-10-15 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-args.m4:
+ add some .m4's for argument checking that can be shared among modules
+
+2005-10-15 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/as-libtool.m4:
+ set _LT_LDFLAGS
+ * m4/gst-libxml2.m4:
+ document
+
+2005-10-15 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-arch.m4:
+ indent a little
+ add AC_REQUIRE
+ * m4/gst-error.m4:
+ clean up
+
+2005-10-12 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gst-autogen.sh:
+ update version detection expression to catch stuff like
+ Libtool (libtool15) 1.5.0
+
+2005-10-11 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gst.supp:
+ commit 6 new suppressions related to g_module_open; can these
+ really not be folded into one ?
+
+2005-10-11 Edward Hervey <edward@fluendo.com>
+
+ * gst.supp:
+ made the <g_type_init calloc 2> suppression more generic
+ Added pthread memleak suppresions
+ Added nss_parse_* memleak suppresion (used by g_option_context_parse)
+
+2005-10-11 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * check.mak:
+ be more strict, more leak resolution
+ * gst.supp:
+ clean up the g_type_init suppressions
+
+2005-10-07 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/Makefile.am:
+ * m4/gst-valgrind.m4:
+ put the valgrind detection in an .m4
+
+2005-09-29 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * check.mak:
+ add some more targets, like "help", but also more intensive tests
+
+2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gtk-doc.mak:
+ make certain doc warnings fatal so people maintain docs again
+
+2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * Makefile.am:
+ * gtk-doc-plugins.mak:
+ * scangobj-merge.py:
+ merge additions from the .signals.new and .args.new file in
+ the original ones, only updating if necessary
+
+2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gst-xmlinspect.py:
+ * gstdoc-scangobj:
+ * gtk-doc-plugins.mak:
+ fix properly for new API; make update in plugins dir now works
+
+2005-09-20 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gst-xmlinspect.py:
+ * gstdoc-scangobj:
+ some fixes for new API
+ * gtk-doc-plugins.mak:
+ set environment properly
+
+2005-09-17 David Schleef <ds@schleef.org>
+
+ * gtk-doc-plugins.mak: Use new environment variables.
+
+2005-09-16 Michael Smith <msmith@fluendo.com>
+
+ * gstdoc-scangobj:
+ Make the scanobj code reflect registry/plugin API changes
+
+2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gtk-doc-plugins.mak:
+ split out scanobj step (which will be run by doc maintainer)
+ from scan step (which will be run on every build)
+ clean up some of the commands for make distcheck
+
+2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gtk-doc-plugins.mak:
+ * mangle-tmpl.py:
+ first stab at reorganizing the plugins build so we can maintain
+ element docs
+
+2005-09-14 David Schleef <ds@schleef.org>
+
+ * as-libtool.mak: Remove
+ * m4/as-libtool.m4: The libtool bug that this worked around has
+ been fixed.
+ * m4/as-version.m4: Don't define GST_RELEASE, since it causes
+ config.h to be regenerated needlessly, and we don't use it.
+
+2005-09-14 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gtk-doc-plugins.mak:
+ error out on inspect failure
+
+2005-09-14 Michael Smith <msmith@fluendo.com>
+
+ * glib-gen.mak:
+ Don't call glib-mkenums with arguments that confuse/break MinGW,
+ fixes 316155.
+
+2005-09-03 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gtk-doc-plugins.mak:
+ * gtk-doc.mak:
+ * m4/gst-doc.m4:
+ separate out gtk-doc and docbook stuff
+ have two separate --enable configure flags
+
+2005-08-26 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * check.mak:
+ add a .gdb target; rebuild registry for each target, otherwise
+ a code rebuild always triggers a reg rebuild, and it's just too
+ annoying
+ * gstdoc-scangobj:
+
+2005-08-21 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * check.mak:
+ separate out REGISTRY_ENVIRONMENT; we want to use that from
+ our valgrind runs, but we also want TESTS_ENVIRONMENT to contain
+ everything that the first test, gst-register, needs
+
+2005-08-21 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * check.mak:
+ parse output of valgrind and check for definitely lost, and error
+ out; somehow I was led to believe valgrind returns non-zero for
+ leaks, but I can't make it do that, so let's parse
+
+2005-08-20 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * check.mak:
+ for some weird reason valgrind does not report actual memleaks
+ if GST_PLUGIN_PATH is set to anything but the core gstreamer dir
+ while valgrind is running. Since the registry is going to go
+ anyway, I don't want to waste any more time on this; I just run
+ valgrind without GST_PLUGIN_PATH set. Since the registry loading
+ doesn't check if GST_PLUGIN_PATH got changed as a reason to rebuild
+ the registry, that's actually fine.
+
+2005-08-15 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * mangle-tmpl.py:
+ keep original Long_Description; only insert an include if it's
+ not already the first line in there
+ * plugins.xsl:
+ output more information for plugins, including an origin hyperlink
+
+2005-08-15 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gst-xmlinspect.py:
+ a first stab at inspecting plugins and outputting an xml description
+ * gtk-doc-plugins.mak:
+ a gtk-doc using snippet for plugins documentation
+ * plugins.xsl:
+ a stylesheet to convert gst-xmlinspect.py output to docbook output
+ for inclusion in the gtk-doc stuff
+
+2005-07-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+
+ * m4/gst-doc.m4:
+ s/pdf/eps/ in test for whether we output EPS images (#309379).
+
+2005-07-18 Andy Wingo <wingo@pobox.com>
+
+ * m4/as-libtool-tags.m4: Ooh, backported from libtool 1.6. Much
+ better. Thanks, Paolo Bonzini!
+
+ * m4/Makefile.am (EXTRA_DIST):
+ * m4/as-libtool-tags.m4: New file, tries to disable some CXX and
+ fortran checks.
+
+2005-07-08 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-error.m4:
+ add macro to set ERROR_CFLAGS
+
+2005-06-30 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * gst-autogen.sh:
+ Remove the old autoregen.sh if it exists before recreating it,
+ to prevent confusing any shell process that might be reading it
+ currently.
+
+2005-06-29 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gtk-doc.m4:
+ added
+
+2005-06-03 Stefan Kost <ensonic@users.sf.net>
+
+ * gst-autogen.sh: create autoregen.sh *before* shifting the options
+
+2005-05-17 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gst-autogen.sh: only update autoregen.sh on actual runs
+
+2005-03-11 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/check.m4: m4 from the check unit test suite
+
+2004-12-14 David Schleef <ds@schleef.org>
+
+ * m4/gst-arch.m4: remove MMX stuff, since it doesn't work and
+ isn't needed anywhere
+
+2004-12-08 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gst-autogen.sh:
+ allow failure command to be run so we can clean upfrom autopoint
+
+2004-09-03 Zeeshan Ali Khattak <zeenix@gmail.com>
+ * m4/gst-feature.m4: Trying to correct the GST_CHECK_CONFIGPROG macro
+
+2004-07-21 Benjamin Otte <otte@gnome.org>
+
+ * m4/.cvsignore: exciting updates for libtool m4 files
+
+2004-07-12 David Schleef <ds@schleef.org>
+
+ * m4/as-objc.m4: Add a macro to test for objective C
+
+2004-06-12 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-feature.m4:
+ not all of them support --plugin-libs, so redirect stderr
+
+2004-06-12 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/as-scrub-include.m4:
+ sync with upstream to 0.1.4. Fixes #132440
+
+2004-06-07 Benjamin Otte <otte@gnome.org>
+
+ * m4/gst-feature.m4:
+ write a big marker into configure output when checking next plugin
+ to allow easier parsing of why plugins are(n't) built.
+
+2004-06-01 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/as-compiler-flag.m4:
+ * m4/as-compiler.m4:
+ * m4/as-libtool.m4:
+ * m4/as-version.m4:
+ sync with upstream, change sticky options to -ko
+
+2004-05-24 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/as-scrub-include.m4: synced with upstream
+
+2004-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * po.mak:
+ snippet for updating .po files
+
+2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * Makefile.am:
+ * m4/Makefile.am:
+ integrate these with the dist
+
+2004-03-17 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * release.mak: add a release target
+
+2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ patch by: Stephane Loeuillet
+
+ * m4/ax_create_stdint_h.m4:
+ use head -n instead of head - (#136500)
+
+2004-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-doc.m4: don't build PS without dvips binary
+
+2004-02-22 Julio M. Merino Vidal <jmmv@menta.net>
+
+ reviewed by: Benjamin Otte <otte@gnome.org>
+
+ * m4/as-docbook.m4:
+ don't use == operator with test(1) (fixes #135115)
+
+2004-02-16 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * common/m4/gst-arch.m4: x86_64 is x86 too (clue from Fedora 2 test)
+
+2004-02-13 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-feature.m4:
+ remove AM_CONDITIONAL for the subsystem since automake 1.6.x
+ requires that call be in configure.ac
+
+2004-02-13 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-libxml2.m4:
+ take required version as argument, and default to 2.4.9 if not
+ specified
+
+2004-02-12 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/gst-feature.m4:
+ rename and fix up GST_CHECK_DISABLE_SUBSYSTEM
+
+2004-02-11 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * common/m4/as-ac-expand.m4:
+ * common/m4/as-auto-alt.m4:
+ * common/m4/as-compiler-flag.m4:
+ * common/m4/as-compiler.m4:
+ * common/m4/as-docbook.m4:
+ * common/m4/as-libtool.m4:
+ * common/m4/as-scrub-include.m4:
+ * common/m4/as-version.m4:
+ * common/m4/glib-gettext.m4:
+ * common/m4/gst-arch.m4:
+ * common/m4/gst-debuginfo.m4:
+ * common/m4/gst-doc.m4:
+ * common/m4/gst-feature.m4:
+ * common/m4/gst-function.m4:
+ * common/m4/gst-glib2.m4:
+ * common/m4/gst-gstreamer.m4:
+ * common/m4/gst-libxml2.m4:
+ * common/m4/gst-makecontext.m4:
+ * common/m4/gst-mcsc.m4:
+ * common/m4/pkg.m4:
+ fix underquoted macros as reported by automake 1.8.x (#133800)
+
+2004-02-11 Johan Dahlin <johan@gnome.org>
+
+ * gst-autogen.sh: Use A-Z instead of A-z in sed expression to
+ avoid a warning
+
+2004-02-05 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+
+ * m4/gst-doc.m4:
+ we use --output-format=xml and --ingnore-files options to
+ gtkdoc-mkdb, which got added between 0.9 and 1.0
+
+2004-02-04 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * m4/as-libtool.m4: remove AM_PROG_LIBTOOL so it can move back
+ to configure.ac to shut up libtoolize
+
+2004-02-03 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * glib-gen.mak: added; used to generate enums and marshal code
+
+2004-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gettext.patch: added; used by autogen.sh to make sure
+ GETTEXT_PACKAGE is understood from po/Makefile.in.in -> po/Makefile.in
+
--- /dev/null
+SUBDIRS = m4
+
+EXTRA_DIST = \
+ ChangeLog \
+ gettext.patch \
+ glib-gen.mak gtk-doc.mak upload-doc.mak \
+ cruft.mak release.mak win32.mak po.mak \
+ parallel-subdirs.mak \
+ gst-autogen.sh \
+ check-exports \
+ c-to-xml.py mangle-tmpl.py scangobj-merge.py \
+ gtk-doc-plugins.mak \
+ plugins.xsl gstdoc-scangobj \
+ gst.supp check.mak \
+ coverage/lcov.mak \
+ coverage/coverage-report.pl \
+ coverage/coverage-report.xsl \
+ coverage/coverage-report-entry.pl \
+ download-translations \
+ extract-release-date-from-doap-file \
+ gst-indent \
+ orc.mak
--- /dev/null
+# Makefile.in generated by automake 1.14.1 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994-2013 Free Software Foundation, Inc.
+
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
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+ fi; \
+ fi
+ctags: ctags-recursive
+
+CTAGS: ctags
+ctags-am: $(TAGS_DEPENDENCIES) $(am__tagged_files)
+ $(am__define_uniq_tagged_files); \
+ test -z "$(CTAGS_ARGS)$$unique" \
+ || $(CTAGS) $(CTAGSFLAGS) $(AM_CTAGSFLAGS) $(CTAGS_ARGS) \
+ $$unique
+
+GTAGS:
+ here=`$(am__cd) $(top_builddir) && pwd` \
+ && $(am__cd) $(top_srcdir) \
+ && gtags -i $(GTAGS_ARGS) "$$here"
+cscopelist: cscopelist-recursive
+
+cscopelist-am: $(am__tagged_files)
+ list='$(am__tagged_files)'; \
+ case "$(srcdir)" in \
+ [\\/]* | ?:[\\/]*) sdir="$(srcdir)" ;; \
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+ esac; \
+ for i in $$list; do \
+ if test -f "$$i"; then \
+ echo "$(subdir)/$$i"; \
+ else \
+ echo "$$sdir/$$i"; \
+ fi; \
+ done >> $(top_builddir)/cscope.files
+
+distclean-tags:
+ -rm -f TAGS ID GTAGS GRTAGS GSYMS GPATH tags
+
+distdir: $(DISTFILES)
+ @srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+ topsrcdirstrip=`echo "$(top_srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+ list='$(DISTFILES)'; \
+ dist_files=`for file in $$list; do echo $$file; done | \
+ sed -e "s|^$$srcdirstrip/||;t" \
+ -e "s|^$$topsrcdirstrip/|$(top_builddir)/|;t"`; \
+ case $$dist_files in \
+ */*) $(MKDIR_P) `echo "$$dist_files" | \
+ sed '/\//!d;s|^|$(distdir)/|;s,/[^/]*$$,,' | \
+ sort -u` ;; \
+ esac; \
+ for file in $$dist_files; do \
+ if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+ if test -d $$d/$$file; then \
+ dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+ if test -d "$(distdir)/$$file"; then \
+ find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+ fi; \
+ if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+ cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+ find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+ fi; \
+ cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+ else \
+ test -f "$(distdir)/$$file" \
+ || cp -p $$d/$$file "$(distdir)/$$file" \
+ || exit 1; \
+ fi; \
+ done
+ @list='$(DIST_SUBDIRS)'; for subdir in $$list; do \
+ if test "$$subdir" = .; then :; else \
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+ || test -d "$(distdir)/$$subdir" \
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+ $(am__relativize); \
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+ @$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
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+ $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
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+ install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+ "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+ fi
+mostlyclean-generic:
+
+clean-generic:
+
+distclean-generic:
+ -test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+ -test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+ @echo "This command is intended for maintainers to use"
+ @echo "it deletes files that may require special tools to rebuild."
+clean: clean-recursive
+
+clean-am: clean-generic clean-libtool mostlyclean-am
+
+distclean: distclean-recursive
+ -rm -f Makefile
+distclean-am: clean-am distclean-generic distclean-tags
+
+dvi: dvi-recursive
+
+dvi-am:
+
+html: html-recursive
+
+html-am:
+
+info: info-recursive
+
+info-am:
+
+install-data-am:
+
+install-dvi: install-dvi-recursive
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-recursive
+
+install-html-am:
+
+install-info: install-info-recursive
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-recursive
+
+install-pdf-am:
+
+install-ps: install-ps-recursive
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-recursive
+ -rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-recursive
+
+mostlyclean-am: mostlyclean-generic mostlyclean-libtool
+
+pdf: pdf-recursive
+
+pdf-am:
+
+ps: ps-recursive
+
+ps-am:
+
+uninstall-am:
+
+.MAKE: $(am__recursive_targets) install-am install-strip
+
+.PHONY: $(am__recursive_targets) CTAGS GTAGS TAGS all all-am check \
+ check-am clean clean-generic clean-libtool cscopelist-am ctags \
+ ctags-am distclean distclean-generic distclean-libtool \
+ distclean-tags distdir dvi dvi-am html html-am info info-am \
+ install install-am install-data install-data-am install-dvi \
+ install-dvi-am install-exec install-exec-am install-html \
+ install-html-am install-info install-info-am install-man \
+ install-pdf install-pdf-am install-ps install-ps-am \
+ install-strip installcheck installcheck-am installdirs \
+ installdirs-am maintainer-clean maintainer-clean-generic \
+ mostlyclean mostlyclean-generic mostlyclean-libtool pdf pdf-am \
+ ps ps-am tags tags-am uninstall uninstall-am
+
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
--- /dev/null
+GStreamer @SERIES_VERSION@
+
+WHAT IT IS
+----------
+
+This is GStreamer, a framework for streaming media.
+
+WHERE TO START
+--------------
+
+We have a website at
+http://gstreamer.freedesktop.org/
+
+You should start by going through our FAQ at
+http://gstreamer.freedesktop.org/data/doc/gstreamer/head/faq/html/
+
+There is more documentation; go to
+http://gstreamer.freedesktop.org/documentation
+
+You can subscribe to our mailing lists; see the website for details.
+
+We track bugs in GNOME's bugzilla; see the website for details.
+
+You can join us on IRC - #gstreamer on irc.freenode.org
+
+GStreamer 1.0 series
+--------------------
+
+Starring
+
+ GSTREAMER
+
+The core around which all other modules revolve. Base functionality and
+libraries, some essential elements, documentation, and testing.
+
+ BASE
+
+A well-groomed and well-maintained collection of GStreamer plug-ins and
+elements, spanning the range of possible types of elements one would want
+to write for GStreamer.
+
+And introducing, for the first time ever, on the development screen ...
+
+ THE GOOD
+
+ --- "Such ingratitude. After all the times I've saved your life."
+
+A collection of plug-ins you'd want to have right next to you on the
+battlefield. Shooting sharp and making no mistakes, these plug-ins have it
+all: good looks, good code, and good licensing. Documented and dressed up
+in tests. If you're looking for a role model to base your own plug-in on,
+here it is.
+
+If you find a plot hole or a badly lip-synced line of code in them,
+let us know - it is a matter of honour for us to ensure Blondie doesn't look
+like he's been walking 100 miles through the desert without water.
+
+ THE UGLY
+
+ --- "When you have to shoot, shoot. Don't talk."
+
+There are times when the world needs a color between black and white.
+Quality code to match the good's, but two-timing, backstabbing and ready to
+sell your freedom down the river. These plug-ins might have a patent noose
+around their neck, or a lock-up license, or any other problem that makes you
+think twice about shipping them.
+
+We don't call them ugly because we like them less. Does a mother love her
+son less because he's not as pretty as the other ones ? No - she commends
+him on his great personality. These plug-ins are the life of the party.
+And we'll still step in and set them straight if you report any unacceptable
+behaviour - because there are two kinds of people in the world, my friend:
+those with a rope around their neck and the people who do the cutting.
+
+ THE BAD
+
+ --- "That an accusation?"
+
+No perfectly groomed moustache or any amount of fine clothing is going to
+cover up the truth - these plug-ins are Bad with a capital B.
+They look fine on the outside, and might even appear to get the job done, but
+at the end of the day they're a black sheep. Without a golden-haired angel
+to watch over them, they'll probably land in an unmarked grave at the final
+showdown.
+
+Don't bug us about their quality - exercise your Free Software rights,
+patch up the offender and send us the patch on the fastest steed you can
+steal from the Confederates. Because you see, in this world, there's two
+kinds of people, my friend: those with loaded guns and those who dig.
+You dig.
+
+The Lowdown
+-----------
+
+ --- "I've never seen so many plug-ins wasted so badly."
+
+GStreamer Plug-ins has grown so big that it's hard to separate the wheat from
+the chaff. Also, distributors have brought up issues about the legal status
+of some of the plug-ins we ship. To remedy this, we've divided the previous
+set of available plug-ins into four modules:
+
+- gst-plugins-base: a small and fixed set of plug-ins, covering a wide range
+ of possible types of elements; these are continuously kept up-to-date
+ with any core changes during the development series.
+
+ - We believe distributors can safely ship these plug-ins.
+ - People writing elements should base their code on these elements.
+ - These elements come with examples, documentation, and regression tests.
+
+- gst-plugins-good: a set of plug-ins that we consider to have good quality
+ code, correct functionality, our preferred license (LGPL for the plug-in
+ code, LGPL or LGPL-compatible for the supporting library).
+
+ - We believe distributors can safely ship these plug-ins.
+ - People writing elements should base their code on these elements.
+
+- gst-plugins-ugly: a set of plug-ins that have good quality and correct
+ functionality, but distributing them might pose problems. The license
+ on either the plug-ins or the supporting libraries might not be how we'd
+ like. The code might be widely known to present patent problems.
+
+ - Distributors should check if they want/can ship these plug-ins.
+ - People writing elements should base their code on these elements.
+
+- gst-plugins-bad: a set of plug-ins that aren't up to par compared to the
+ rest. They might be close to being good quality, but they're missing
+ something - be it a good code review, some documentation, a set of tests,
+ a real live maintainer, or some actual wide use.
+ If the blanks are filled in they might be upgraded to become part of
+ either gst-plugins-good or gst-plugins-ugly, depending on the other factors.
+
+ - If the plug-ins break, you can't complain - instead, you can fix the
+ problem and send us a patch, or bribe someone into fixing them for you.
+ - New contributors can start here for things to work on.
+
+PLATFORMS
+---------
+
+- Linux is of course fully supported
+- FreeBSD is reported to work; other BSDs should work too
+- Solaris is reported to work; a specific sunaudiosink plugin has been written
+- MacOSX works, binary 1.x packages can be built using the cerbero build tool
+- Windows works; binary 1.x packages can be built using the cerbero build tool
+ - MSys/MinGW builds
+ - Microsoft Visual Studio builds are not yet available or supported
+- Android works, binary 1.x packages can be built using the cerbero build tool
+- iOS works
+
+INSTALLING FROM PACKAGES
+------------------------
+
+You should always prefer installing from packages first. GStreamer is
+well-maintained for a number of distributions, including Fedora, Debian,
+Ubuntu, Mandrake, Gentoo, ...
+
+Only in cases where you:
+- want to hack on GStreamer
+- want to verify that a bug has been fixed
+- do not have a sane distribution
+should you choose to build from source tarballs or git.
+
+Find more information about the various packages at
+http://gstreamer.freedesktop.org/download/
+
+COMPILING FROM SOURCE TARBALLS
+------------------------------
+
+- again, make sure that you really need to install from source !
+ If GStreamer is one of your first projects ever that you build from source,
+ consider taking on an easier project.
+
+- check output of ./configure --help to see if any options apply to you
+- run
+ ./configure
+ make
+
+ to build GStreamer.
+- if you want to install it (not required, but what you usually want to do), run
+ make install
+
+- try out a simple test:
+ gst-launch -v fakesrc num_buffers=5 ! fakesink
+ (If you didn't install GStreamer, prefix gst-launch with tools/)
+
+ If it outputs a bunch of messages from fakesrc and fakesink, everything is
+ ok.
+
+ If it did not work, keep in mind that you might need to adjust the
+ PATH and/or LD_LIBRARY_PATH environment variables to make the system
+ find GStreamer in the prefix where you installed (by default that is /usr/local).
+
+- After this, you're ready to install gst-plugins, which will provide the
+ functionality you're probably looking for by now, so go on and read
+ that README.
+
+COMPILING FROM GIT
+------------------
+
+When building from git sources, you will need to run autogen.sh to generate
+the build system files.
+
+You will need a set of additional tools typical for building from git,
+including:
+- autoconf
+- automake
+- libtool
+
+autogen.sh will check for recent enough versions and complain if you don't have
+them. You can also specify specific versions of automake and autoconf with
+--with-automake and --with-autoconf
+
+Check autogen.sh options by running autogen.sh --help
+
+autogen.sh can pass on arguments to configure
+
+When you have done this once, you can use autoregen.sh to re-autogen with
+the last passed options as a handy shortcut. Use it.
+
+After the autogen.sh stage, you can follow the directions listed in
+"COMPILING FROM SOURCE"
+
+You can also run your whole git stack uninstalled in your home directory,
+so that you can quickly test changes without affecting your system setup or
+interfering with GStreamer installed from packages. Many GStreamer developers
+use an uninstalled setup for their work.
+
+There is a 'create-uninstalled-setup.sh' script in
+
+ http://cgit.freedesktop.org/gstreamer/gstreamer/tree/scripts/
+
+to easily create an uninstalled setup from scratch.
+
+
+PLUG-IN DEPENDENCIES AND LICENSES
+---------------------------------
+
+GStreamer is developed under the terms of the LGPL (see LICENSE file for
+details). Some of our plug-ins however rely on libraries which are available
+under other licenses. This means that if you are distributing an application
+which has a non-GPL compatible license (for instance a closed-source
+application) with GStreamer, you have to make sure not to distribute GPL-linked
+plug-ins.
+
+When using GPL-linked plug-ins, GStreamer is for all practical reasons
+under the GPL itself.
+
+HISTORY
+-------
+
+The fundamental design comes from the video pipeline at Oregon Graduate
+Institute, as well as some ideas from DirectMedia. It's based on plug-ins that
+will provide the various codec and other functionality. The interface
+hopefully is generic enough for various companies (ahem, Apple) to release
+binary codecs for Linux, until such time as they get a clue and release the
+source.
--- /dev/null
+# -*- Mode: Python -*-
+# vi:si:et:sw=4:sts=4:ts=4
+
+"""
+Convert a C program to valid XML to be included in docbook
+"""
+
+from __future__ import print_function, unicode_literals
+
+import sys
+import os
+from xml.sax import saxutils
+
+def main():
+ if len(sys.argv) == 1:
+ sys.stderr.write("Please specify a source file to convert")
+ sys.exit(1)
+ source = sys.argv[1]
+
+ if not os.path.exists(source):
+ sys.stderr.write("%s does not exist.\n" % source)
+ sys.exit(1)
+
+ content = open(source, "r").read()
+
+ # print header
+ print ('<?xml version="1.0"?>')
+ print ('<!DOCTYPE refentry PUBLIC "-//OASIS//DTD DocBook XML V4.1.2//EN" "http://www.oasis-open.org/docbook/xml/4.1.2/docbookx.dtd">')
+ print ()
+ print ('<programlisting>')
+
+ # print content
+ print (saxutils.escape(content))
+ print ('</programlisting>')
+
+main()
--- /dev/null
+#!/bin/sh
+# check-exports
+#
+# quick'n'dirty script that retrieves the list of exported symbols of a given
+# library using 'nm', and compares that against the list of symbols-to-export
+# of our win32/common/libfoo.def files.
+
+if [ $# -ne 2 ]; then
+ echo "Usage: $0 library.def library.so"
+ exit 1
+fi
+
+def_path="$1"
+def_name="$(basename $def_path)"
+lib_path="$2"
+
+lib_result="`mktemp /tmp/defname.XXXXXX`"
+
+LC_ALL=C
+export LC_ALL
+
+# On Solaris, add -p to get the correct output format
+NMARGS=
+if nm -V 2>&1 |grep Solaris > /dev/null; then
+ NMARGS=-p
+fi
+
+# _end is special cased because for some reason it is reported as an exported
+# BSS symbol, unlike on linux where it's a local absolute symbol.
+nm $NMARGS $lib_path | awk \
+ '{
+ if ($3 ~ /^[_]?(gst_|Gst|GST_).*/)
+ {
+ if ($2 ~ /^[BSDG]$/)
+ print "\t" $3 " DATA"
+ else if ($2 == "T")
+ print "\t" $3
+ }
+ }' | sort | awk '{ if (NR == 1) print "EXPORTS"; print $0; }' \
+ > $lib_result
+
+diffoutput=`diff -u $def_path $lib_result`
+diffresult=$?
+
+rm $lib_result
+
+if test "$diffresult" -eq 0; then
+ exit 0;
+else
+ echo -n "$diffoutput" >&2
+ echo >&2
+ exit 1;
+fi
+
--- /dev/null
+# keep target around, since it's referenced in the modules' Makefiles
+clean-local-check:
+ @echo
+
+if HAVE_VALGRIND
+# hangs spectacularly on some machines, so let's not do this by default yet
+check-valgrind:
+ $(MAKE) valgrind
+else
+check-valgrind:
+ @true
+endif
+
+LOOPS ?= 10
+
+# run any given test by running make test.check
+# if the test fails, run it again at at least debug level 2
+%.check: %
+ @$(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=20 \
+ $* || \
+ $(TESTS_ENVIRONMENT) \
+ GST_DEBUG=$$GST_DEBUG,*:2 \
+ CK_DEFAULT_TIMEOUT=20 \
+ $*
+
+# just like 'check', but don't run it again if it fails (useful for debugging)
+%.check-norepeat: %
+ @$(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=20 \
+ $*
+
+# run any given test in a loop
+%.torture: %
+ @for i in `seq 1 $(LOOPS)`; do \
+ $(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=20 \
+ $*; done
+
+# run any given test in an infinite loop
+%.forever: %
+ @while true; do \
+ $(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=20 \
+ $* || break; done
+
+# valgrind any given test by running make test.valgrind
+%.valgrind: %
+ @$(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=360 \
+ G_SLICE=always-malloc \
+ $(LIBTOOL) --mode=execute \
+ $(VALGRIND_PATH) -q \
+ $(foreach s,$(SUPPRESSIONS),--suppressions=$(s)) \
+ --tool=memcheck --leak-check=full --trace-children=yes \
+ --show-possibly-lost=no \
+ --leak-resolution=high --num-callers=20 \
+ ./$* 2>&1 | tee valgrind.log
+ @if grep "==" valgrind.log > /dev/null 2>&1; then \
+ rm valgrind.log; \
+ exit 1; \
+ fi
+ @rm valgrind.log
+
+# valgrind any given test and generate suppressions for it
+%.valgrind.gen-suppressions: %
+ @$(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=360 \
+ G_SLICE=always-malloc \
+ $(LIBTOOL) --mode=execute \
+ $(VALGRIND_PATH) -q \
+ $(foreach s,$(SUPPRESSIONS),--suppressions=$(s)) \
+ --tool=memcheck --leak-check=full --trace-children=yes \
+ --show-possibly-lost=no \
+ --leak-resolution=high --num-callers=20 \
+ --gen-suppressions=all \
+ ./$* 2>&1 | tee suppressions.log
+
+# valgrind torture any given test
+%.valgrind-torture: %
+ @for i in `seq 1 $(LOOPS)`; do \
+ $(MAKE) $*.valgrind || \
+ (echo "Failure after $$i runs"; exit 1) || \
+ exit 1; \
+ done
+ @banner="All $(LOOPS) loops passed"; \
+ dashes=`echo "$$banner" | sed s/./=/g`; \
+ echo $$dashes; echo $$banner; echo $$dashes
+
+# valgrind any given test until failure by running make test.valgrind-forever
+%.valgrind-forever: %
+ @while $(MAKE) $*.valgrind; do \
+ true; done
+
+# gdb any given test by running make test.gdb
+%.gdb: %
+ @$(TESTS_ENVIRONMENT) \
+ CK_FORK=no \
+ $(LIBTOOL) --mode=execute \
+ gdb $*
+
+%.lcov-reset:
+ $(MAKE) $*.lcov-run
+ $(MAKE) $*.lcov-report
+
+%.lcov: %
+ $(MAKE) $*.lcov-reset
+
+if GST_GCOV_ENABLED
+%.lcov-clean:
+ $(MAKE) -C $(top_builddir) lcov-clean
+
+%.lcov-run:
+ $(MAKE) $*.lcov-clean
+ $(MAKE) $*.check
+
+%.lcov-report:
+ $(MAKE) -C $(top_builddir) lcov-report
+else
+%.lcov-run:
+ echo "Need to reconfigure with --enable-gcov"
+
+%.lcov-report:
+ echo "Need to reconfigure with --enable-gcov"
+endif
+
+# torture tests
+torture: $(TESTS)
+ -rm test-registry.*
+ @echo "Torturing tests ..."
+ @for i in `seq 1 $(LOOPS)`; do \
+ $(MAKE) check || \
+ (echo "Failure after $$i runs"; exit 1) || \
+ exit 1; \
+ done
+ @banner="All $(LOOPS) loops passed"; \
+ dashes=`echo "$$banner" | sed s/./=/g`; \
+ echo $$dashes; echo $$banner; echo $$dashes
+
+# forever tests
+forever: $(TESTS)
+ -rm test-registry.*
+ @echo "Forever tests ..."
+ @while true; do \
+ $(MAKE) check || \
+ (echo "Failure"; exit 1) || \
+ exit 1; \
+ done
+
+# valgrind all tests
+valgrind: $(TESTS)
+ @echo "Valgrinding tests ..."
+ @failed=0; \
+ for t in $(filter-out $(VALGRIND_TESTS_DISABLE),$(TESTS)); do \
+ $(MAKE) $$t.valgrind; \
+ if test "$$?" -ne 0; then \
+ echo "Valgrind error for test $$t"; \
+ failed=`expr $$failed + 1`; \
+ whicht="$$whicht $$t"; \
+ fi; \
+ done; \
+ if test "$$failed" -ne 0; then \
+ echo "$$failed tests had leaks or errors under valgrind:"; \
+ echo "$$whicht"; \
+ false; \
+ fi
+
+# valgrind all tests until failure
+valgrind-forever: $(TESTS)
+ -rm test-registry.*
+ @echo "Forever valgrinding tests ..."
+ @while true; do \
+ $(MAKE) valgrind || \
+ (echo "Failure"; exit 1) || \
+ exit 1; \
+ done
+
+# valgrind torture all tests
+valgrind-torture: $(TESTS)
+ -rm test-registry.*
+ @echo "Torturing and valgrinding tests ..."
+ @for i in `seq 1 $(LOOPS)`; do \
+ $(MAKE) valgrind || \
+ (echo "Failure after $$i runs"; exit 1) || \
+ exit 1; \
+ done
+ @banner="All $(LOOPS) loops passed"; \
+ dashes=`echo "$$banner" | sed s/./=/g`; \
+ echo $$dashes; echo $$banner; echo $$dashes
+
+# valgrind all tests and generate suppressions
+valgrind.gen-suppressions: $(TESTS)
+ @echo "Valgrinding tests ..."
+ @failed=0; \
+ for t in $(filter-out $(VALGRIND_TESTS_DISABLE),$(TESTS)); do \
+ $(MAKE) $$t.valgrind.gen-suppressions; \
+ if test "$$?" -ne 0; then \
+ echo "Valgrind error for test $$t"; \
+ failed=`expr $$failed + 1`; \
+ whicht="$$whicht $$t"; \
+ fi; \
+ done; \
+ if test "$$failed" -ne 0; then \
+ echo "$$failed tests had leaks or errors under valgrind:"; \
+ echo "$$whicht"; \
+ false; \
+ fi
+
+# inspect every plugin feature
+GST_INSPECT = $(GST_TOOLS_DIR)/gst-inspect-$(GST_API_VERSION)
+inspect:
+ @echo "Inspecting features ..."
+ @for e in `$(TESTS_ENVIRONMENT) $(GST_INSPECT) | head -n -2 \
+ | cut -d: -f2`; \
+ do echo Inspecting $$e; \
+ $(GST_INSPECT) $$e > /dev/null 2>&1; done
+
+help:
+ @echo
+ @echo "make check -- run all checks"
+ @echo "make torture -- run all checks $(LOOPS) times"
+ @echo "make (dir)/(test).check -- run the given check once, repeat with GST_DEBUG=*:2 if it fails"
+ @echo "make (dir)/(test).check-norepeat -- run the given check once, but don't run it again if it fails"
+ @echo "make (dir)/(test).forever -- run the given check forever"
+ @echo "make (dir)/(test).torture -- run the given check $(LOOPS) times"
+ @echo
+ @echo "make (dir)/(test).gdb -- start up gdb for the given test"
+ @echo
+ @echo "make valgrind -- valgrind all tests"
+ @echo "make valgrind-forever -- valgrind all tests forever"
+ @echo "make valgrind-torture -- valgrind all tests $(LOOPS) times"
+ @echo "make valgrind.gen-suppressions -- generate suppressions for all tests"
+ @echo " and save to suppressions.log"
+ @echo "make (dir)/(test).valgrind -- valgrind the given test"
+ @echo "make (dir)/(test).valgrind-forever -- valgrind the given test forever"
+ @echo "make (dir)/(test).valgrind-torture -- valgrind the given test $(LOOPS) times"
+ @echo "make (dir)/(test).valgrind.gen-suppressions -- generate suppressions"
+ @echo " and save to suppressions.log"
+ @echo "make inspect -- inspect all plugin features"
+ @echo
+ @echo
+ @echo "Additionally, you can use the GST_CHECKS environment variable to"
+ @echo "specify which test(s) should be run. This is useful if you are"
+ @echo "debugging a failure in one particular test, or want to reproduce"
+ @echo "a race condition in a single test."
+ @echo
+ @echo "Examples:"
+ @echo
+ @echo " GST_CHECKS=test_this,test_that make element/foobar.check"
+ @echo " GST_CHECKS=test_many_threads make element/foobar.forever"
+ @echo
+
--- /dev/null
+#!/usr/bin/perl
+#
+# Copyright (C) 2006 Daniel Berrange
+#
+# This program is free software; you can redistribute it and/or modify
+# it under the terms of the GNU General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+
+
+print <<EOF;
+<html>
+<head>
+<title>Coverage report for $ARGV[0]</title>
+<style type="text/css">
+ span.perfect {
+ background: rgb(0,255,0);
+ }
+ span.terrible {
+ background: rgb(255,0,0);
+ }
+</style>
+</head>
+<body>
+<h1>Coverage report for $ARGV[0]</h1>
+
+<pre>
+EOF
+
+
+while (<>) {
+ s/&/&/g;
+ s/</</g;
+ s/>/>/g;
+
+ if (/^\s*function (\S+) called (\d+) returned \d+% blocks executed \d+%/) {
+ my $class = $2 > 0 ? "perfect" : "terrible";
+ $_ = "<span class=\"$class\" id=\"" . $1 . "\">$_</span>";
+ } elsif (/^\s*branch\s+\d+\s+taken\s+(\d+)%\s+.*$/) {
+ my $class = $1 > 0 ? "perfect" : "terrible";
+ $_ = "<span class=\"$class\">$_</span>";
+ } elsif (/^\s*branch\s+\d+\s+never executed.*$/) {
+ my $class = "terrible";
+ $_ = "<span class=\"$class\">$_</span>";
+ } elsif (/^\s*call\s+\d+\s+never executed.*$/) {
+ my $class = "terrible";
+ $_ = "<span class=\"$class\">$_</span>";
+ } elsif (/^\s*call\s+\d+\s+returned\s+(\d+)%.*$/) {
+ my $class = $1 > 0 ? "perfect" : "terrible";
+ $_ = "<span class=\"$class\">$_</span>";
+ }
+
+ print;
+}
+
+print <<EOF;
+</pre>
+</body>
+</html>
+EOF
--- /dev/null
+#!/usr/bin/perl
+#
+# Copyright (C) 2006 Daniel Berrange
+#
+# This program is free software; you can redistribute it and/or modify
+# it under the terms of the GNU General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+
+use warnings;
+use strict;
+
+my %coverage = ( functions => {}, files => {} );
+
+my %filemap;
+
+my $type;
+my $name;
+
+my @functions;
+
+while (<>) {
+ if (/^Function '(.*)'\s*$/) {
+ $type = "function";
+ $name = $1;
+ $coverage{$type}->{$name} = {};
+ push @functions, $name;
+ } elsif (/^File '(.*?)'\s*$/) {
+ $type = "file";
+ $name = $1;
+ $coverage{$type}->{$name} = {};
+
+ foreach my $func (@functions) {
+ $coverage{"function"}->{$func}->{file} = $name;
+ }
+ @functions = ();
+ } elsif (/^Lines executed:(.*)%\s*of\s*(\d+)\s*$/) {
+ $coverage{$type}->{$name}->{lines} = $2;
+ $coverage{$type}->{$name}->{linesCoverage} = $1;
+ } elsif (/^Branches executed:(.*)%\s*of\s*(\d+)\s*$/) {
+ $coverage{$type}->{$name}->{branches} = $2;
+ $coverage{$type}->{$name}->{branchesCoverage} = $1;
+ } elsif (/^Taken at least once:(.*)%\s*of\s*(\d+)\s*$/) {
+ $coverage{$type}->{$name}->{conds} = $2;
+ $coverage{$type}->{$name}->{condsCoverage} = $1;
+ } elsif (/^Calls executed:(.*)%\s*of\s*(\d+)\s*$/) {
+ $coverage{$type}->{$name}->{calls} = $2;
+ $coverage{$type}->{$name}->{callsCoverage} = $1;
+ } elsif (/^No branches$/) {
+ $coverage{$type}->{$name}->{branches} = 0;
+ $coverage{$type}->{$name}->{branchesCoverage} = "100.00";
+ $coverage{$type}->{$name}->{conds} = 0;
+ $coverage{$type}->{$name}->{condsCoverage} = "100.00";
+ } elsif (/^No calls$/) {
+ $coverage{$type}->{$name}->{calls} = 0;
+ $coverage{$type}->{$name}->{callsCoverage} = "100.00";
+ } elsif (/^\s*(.*):creating '(.*)'\s*$/) {
+ $filemap{$1} = $2;
+ } elsif (/^\s*$/) {
+ # nada
+ } else {
+ warn "Shit [$_]\n";
+ }
+}
+
+my %summary;
+foreach my $type ("function", "file") {
+ $summary{$type} = {};
+ foreach my $m ("lines", "branches", "conds", "calls") {
+ my $totalGot = 0;
+ my $totalMiss = 0;
+ my $count = 0;
+ foreach my $func (keys %{$coverage{function}}) {
+ $count++;
+ my $got = $coverage{function}->{$func}->{$m};
+ $totalGot += $got;
+ my $miss = $got * $coverage{function}->{$func}->{$m ."Coverage"} / 100;
+ $totalMiss += $miss;
+ }
+ $summary{$type}->{$m} = sprintf("%d", $totalGot);
+ $summary{$type}->{$m . "Coverage"} = sprintf("%.2f", $totalMiss / $totalGot * 100);
+ }
+}
+
+
+
+print "<coverage>\n";
+
+foreach my $type ("function", "file") {
+ printf "<%ss>\n", $type;
+ foreach my $name (sort { $a cmp $b } keys %{$coverage{$type}}) {
+ my $rec = $coverage{$type}->{$name};
+ printf " <entry name=\"%s\" details=\"%s\">\n", $name, ($type eq "file" ? $filemap{$name} : $filemap{$rec->{file}});
+ printf " <lines count=\"%s\" coverage=\"%s\"/>\n", $rec->{lines}, $rec->{linesCoverage};
+ if (exists $rec->{branches}) {
+ printf " <branches count=\"%s\" coverage=\"%s\"/>\n", $rec->{branches}, $rec->{branchesCoverage};
+ }
+ if (exists $rec->{conds}) {
+ printf " <conditions count=\"%s\" coverage=\"%s\"/>\n", $rec->{conds}, $rec->{condsCoverage};
+ }
+ if (exists $rec->{calls}) {
+ printf " <calls count=\"%s\" coverage=\"%s\"/>\n", $rec->{calls}, $rec->{callsCoverage};
+ }
+ print " </entry>\n";
+ }
+
+ printf " <summary>\n";
+ printf " <lines count=\"%s\" coverage=\"%s\"/>\n", $summary{$type}->{lines}, $summary{$type}->{linesCoverage};
+ printf " <branches count=\"%s\" coverage=\"%s\"/>\n", $summary{$type}->{branches}, $summary{$type}->{branchesCoverage};
+ printf " <conditions count=\"%s\" coverage=\"%s\"/>\n", $summary{$type}->{conds}, $summary{$type}->{condsCoverage};
+ printf " <calls count=\"%s\" coverage=\"%s\"/>\n", $summary{$type}->{calls}, $summary{$type}->{callsCoverage};
+ printf " </summary>\n";
+ printf "</%ss>\n", $type;
+}
+
+print "</coverage>\n";
--- /dev/null
+<?xml version="1.0" encoding="utf-8"?>\r
+<!--\r
+#\r
+# Copyright (C) 2006 Daniel Berrange\r
+#\r
+# This program is free software; you can redistribute it and/or modify\r
+# it under the terms of the GNU General Public License as published by\r
+# the Free Software Foundation; either version 2 of the License, or\r
+# (at your option) any later version.\r
+#\r
+# This program is distributed in the hope that it will be useful,\r
+# but WITHOUT ANY WARRANTY; without even the implied warranty of\r
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the\r
+# GNU General Public License for more details.\r
+#\r
+# You should have received a copy of the GNU General Public License\r
+# along with this program; if not, write to the Free Software\r
+# Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA\r
+-->\r
+<xsl:stylesheet xmlns:xsl="http://www.w3.org/1999/XSL/Transform"\r
+ version="1.0">\r
+\r
+ <xsl:output method="html"/>\r
+\r
+ <xsl:template match="coverage">\r
+ <html>\r
+ <head>\r
+ <title>Coverage report</title>\r
+ <style type="text/css">\r
+ tbody tr.odd td.label {\r
+ border-top: 1px solid rgb(128,128,128);\r
+ border-bottom: 1px solid rgb(128,128,128);\r
+ }\r
+ tbody tr.odd td.label {\r
+ background: rgb(200,200,200);\r
+ }\r
+ \r
+ thead, tfoot {\r
+ background: rgb(60,60,60);\r
+ color: white;\r
+ font-weight: bold;\r
+ }\r
+\r
+ tr td.perfect {\r
+ background: rgb(0,255,0);\r
+ color: black;\r
+ }\r
+ tr td.excellant {\r
+ background: rgb(140,255,140);\r
+ color: black;\r
+ }\r
+ tr td.good {\r
+ background: rgb(160,255,0);\r
+ color: black;\r
+ }\r
+ tr td.poor {\r
+ background: rgb(255,160,0);\r
+ color: black;\r
+ }\r
+ tr td.bad {\r
+ background: rgb(255,140,140);\r
+ color: black;\r
+ }\r
+ tr td.terrible {\r
+ background: rgb(255,0,0);\r
+ color: black;\r
+ }\r
+ </style>\r
+ </head>\r
+ <body>\r
+ <h1>Coverage report</h1>\r
+ <xsl:apply-templates/>\r
+ </body>\r
+ </html>\r
+ </xsl:template>\r
+\r
+ <xsl:template match="functions">\r
+ <h2>Function coverage</h2>\r
+ <xsl:call-template name="content">\r
+ <xsl:with-param name="type" select="'function'"/>\r
+ </xsl:call-template>\r
+ </xsl:template>\r
+ \r
+\r
+ <xsl:template match="files">\r
+ <h2>File coverage</h2>\r
+ <xsl:call-template name="content">\r
+ <xsl:with-param name="type" select="'file'"/>\r
+ </xsl:call-template>\r
+ </xsl:template>\r
+\r
+ <xsl:template name="content">\r
+ <xsl:param name="type"/>\r
+ <table>\r
+ <thead>\r
+ <tr>\r
+ <th>Name</th>\r
+ <th>Lines</th>\r
+ <th>Branches</th>\r
+ <th>Conditions</th>\r
+ <th>Calls</th>\r
+ </tr>\r
+ </thead>\r
+ <tbody>\r
+ <xsl:for-each select="entry">\r
+ <xsl:call-template name="entry">\r
+ <xsl:with-param name="type" select="$type"/>\r
+ <xsl:with-param name="class">\r
+ <xsl:choose>\r
+ <xsl:when test="position() mod 2">\r
+ <xsl:text>odd</xsl:text>\r
+ </xsl:when>\r
+ <xsl:otherwise>\r
+ <xsl:text>even</xsl:text>\r
+ </xsl:otherwise>\r
+ </xsl:choose>\r
+ </xsl:with-param>\r
+ </xsl:call-template>\r
+ </xsl:for-each>\r
+ </tbody>\r
+ <tfoot>\r
+ <xsl:for-each select="summary">\r
+ <xsl:call-template name="entry">\r
+ <xsl:with-param name="type" select="'summary'"/>\r
+ <xsl:with-param name="class">\r
+ <xsl:choose>\r
+ <xsl:when test="position() mod 2">\r
+ <xsl:text>odd</xsl:text>\r
+ </xsl:when>\r
+ <xsl:otherwise>\r
+ <xsl:text>even</xsl:text>\r
+ </xsl:otherwise>\r
+ </xsl:choose>\r
+ </xsl:with-param>\r
+ </xsl:call-template>\r
+ </xsl:for-each>\r
+ </tfoot>\r
+ </table>\r
+ </xsl:template>\r
+ \r
+ <xsl:template name="entry">\r
+ <xsl:param name="type"/>\r
+ <xsl:param name="class"/>\r
+ <tr class="{$class}">\r
+ <xsl:choose>\r
+ <xsl:when test="$type = 'function'">\r
+ <td class="label"><a href="{@details}.html#{@name}"><xsl:value-of select="@name"/></a></td>\r
+ </xsl:when>\r
+ <xsl:when test="$type = 'file'">\r
+ <td class="label"><a href="{@details}.html"><xsl:value-of select="@name"/></a></td>\r
+ </xsl:when>\r
+ <xsl:otherwise>\r
+ <td class="label">Summary</td>\r
+ </xsl:otherwise>\r
+ </xsl:choose>\r
+\r
+ <xsl:if test="count(lines)">\r
+ <xsl:apply-templates select="lines"/>\r
+ </xsl:if>\r
+ <xsl:if test="not(count(lines))">\r
+ <xsl:call-template name="missing"/>\r
+ </xsl:if>\r
+\r
+ <xsl:if test="count(branches)">\r
+ <xsl:apply-templates select="branches"/>\r
+ </xsl:if>\r
+ <xsl:if test="not(count(branches))">\r
+ <xsl:call-template name="missing"/>\r
+ </xsl:if>\r
+\r
+ <xsl:if test="count(conditions)">\r
+ <xsl:apply-templates select="conditions"/>\r
+ </xsl:if>\r
+ <xsl:if test="not(count(conditions))">\r
+ <xsl:call-template name="missing"/>\r
+ </xsl:if>\r
+\r
+ <xsl:if test="count(calls)">\r
+ <xsl:apply-templates select="calls"/>\r
+ </xsl:if>\r
+ <xsl:if test="not(count(calls))">\r
+ <xsl:call-template name="missing"/>\r
+ </xsl:if>\r
+\r
+ </tr>\r
+ </xsl:template>\r
+ \r
+ <xsl:template match="lines">\r
+ <xsl:call-template name="row"/>\r
+ </xsl:template>\r
+\r
+ <xsl:template match="branches">\r
+ <xsl:call-template name="row"/>\r
+ </xsl:template>\r
+\r
+ <xsl:template match="conditions">\r
+ <xsl:call-template name="row"/>\r
+ </xsl:template>\r
+\r
+ <xsl:template match="calls">\r
+ <xsl:call-template name="row"/>\r
+ </xsl:template>\r
+\r
+ <xsl:template name="missing">\r
+ <td></td>\r
+ </xsl:template>\r
+\r
+ <xsl:template name="row">\r
+ <xsl:variable name="quality">\r
+ <xsl:choose>\r
+ <xsl:when test="@coverage = 100">\r
+ <xsl:text>perfect</xsl:text>\r
+ </xsl:when>\r
+ <xsl:when test="@coverage >= 80.0">\r
+ <xsl:text>excellant</xsl:text>\r
+ </xsl:when>\r
+ <xsl:when test="@coverage >= 60.0">\r
+ <xsl:text>good</xsl:text>\r
+ </xsl:when>\r
+ <xsl:when test="@coverage >= 40.0">\r
+ <xsl:text>poor</xsl:text>\r
+ </xsl:when>\r
+ <xsl:when test="@coverage >= 20.0">\r
+ <xsl:text>bad</xsl:text>\r
+ </xsl:when>\r
+ <xsl:otherwise>\r
+ <xsl:text>terrible</xsl:text>\r
+ </xsl:otherwise>\r
+ </xsl:choose>\r
+ </xsl:variable>\r
+ \r
+ <td class="{$quality}"><xsl:value-of select="@coverage"/>% of <xsl:value-of select="@count"/></td>\r
+ </xsl:template>\r
+\r
+</xsl:stylesheet>\r
--- /dev/null
+## .PHONY so it always rebuilds it
+.PHONY: lcov-reset lcov lcov-run lcov-report lcov-upload lcov-clean
+
+# run lcov from scratch, always
+lcov-reset:
+ $(MAKE) lcov-run
+ $(MAKE) lcov-report
+
+# run lcov from scratch if the dir is not there
+lcov:
+ $(MAKE) lcov-reset
+
+if GST_GCOV_ENABLED
+# reset lcov stats
+lcov-clean:
+ @-rm -rf lcov
+ lcov --directory . --zerocounters
+
+# reset run coverage tests
+lcov-run:
+ -$(MAKE) lcov-clean
+ -if test -d tests/check; then $(MAKE) -C tests/check inspect; fi
+ -$(MAKE) check
+
+# generate report based on current coverage data
+lcov-report:
+ mkdir lcov
+ lcov --compat-libtool --directory . --capture --output-file lcov/lcov.info
+ lcov --list-full-path -l lcov/lcov.info | grep -v "`cd $(top_srcdir) && pwd`" | cut -d\| -f1 > lcov/remove
+ lcov --list-full-path -l lcov/lcov.info | grep "tests/check/" | cut -d\| -f1 >> lcov/remove
+ lcov --list-full-path -l lcov/lcov.info | grep "docs/plugins/" | cut -d\| -f1 >> lcov/remove
+ lcov -r lcov/lcov.info `cat lcov/remove` > lcov/lcov.cleaned.info
+ rm lcov/remove
+ mv lcov/lcov.cleaned.info lcov/lcov.info
+ genhtml -t "$(PACKAGE_STRING)" -o lcov --num-spaces 2 lcov/lcov.info
+
+lcov-upload: lcov
+ rsync -rvz -e ssh --delete lcov/* gstreamer.freedesktop.org:/srv/gstreamer.freedesktop.org/www/data/coverage/lcov/$(PACKAGE)
+
+else
+lcov-run:
+ echo "Need to reconfigure with --enable-gcov"
+
+lcov-report:
+ echo "Need to reconfigure with --enable-gcov"
+endif
+
--- /dev/null
+# checks for left-over files in the (usually uninstalled) tree, ie. for
+# stuff that best be deleted to avoid problems like having old plugin binaries
+# lying around.
+#
+# set CRUFT_FILES and/or CRUFT_DIRS in your Makefile.am when you include this
+
+check-cruft:
+ @cruft_files=""; cruft_dirs=""; \
+ for f in $(CRUFT_FILES); do \
+ if test -e $$f; then \
+ cruft_files="$$cruft_files $$f"; \
+ fi \
+ done; \
+ for d in $(CRUFT_DIRS); do \
+ if test -e $$d; then \
+ cruft_dirs="$$cruft_dirs $$d"; \
+ fi \
+ done; \
+ if test "x$$cruft_files$$cruft_dirs" != x; then \
+ echo; \
+ echo "**** CRUFT ALERT *****"; \
+ echo; \
+ echo "The following files and directories may not be needed any "; \
+ echo "longer (usually because a plugin has been merged into "; \
+ echo "another plugin, moved to a different module, or been "; \
+ echo "renamed), and you probably want to clean them up if you "; \
+ echo "don't have local changes: "; \
+ echo; \
+ for f in $$cruft_files; do echo "file $$f"; done; \
+ echo; \
+ for d in $$cruft_dirs; do echo "directory $$d"; done; \
+ echo; \
+ echo "'make clean-cruft' will remove these for you."; \
+ echo; \
+ fi
+
+clean-cruft-dirs:
+ @for d in $(CRUFT_DIRS); do \
+ if test -e $$d; then \
+ rm -r "$$d" && echo "Removed directory $$d"; \
+ fi \
+ done
+
+clean-cruft-files:
+ @for f in $(CRUFT_FILES); do \
+ if test -e $$f; then \
+ rm "$$f" && echo "Removed file $$f"; \
+ fi \
+ done
+
+clean-cruft: clean-cruft-dirs clean-cruft-files
+
+# also might want to add this to your Makefile.am:
+#
+# all-local: check-cruft
+
--- /dev/null
+#!/bin/sh
+# Shell script to download the latest translations for a given GStreamer
+# package from translationproject.org
+
+
+# DOMAINS based on http://translationproject.org/extra/matrix.html
+# We need to check all domains, not only po/LINGUAS, since there might be
+# new translations
+DOMAINS=\
+"af am ar az be bg pt_BR bs ca zh_CN cs cy da de el eo es et eu fa fi fr "\
+"ga en_GB gl gu he hi zh_HK hr hu id is it ja ko ku ky lg lt lv mk mn ms "\
+"mt nb ne nl nn or pa pl pt rm ro ru rw sk sl sq sr sv ta tq th tk "\
+"tr zh_TW uk ven vi wa xh zu"
+
+# for testing/debugging:
+#DOMAINS="es fr hu sv pl xx"
+
+# check for 'diff' program
+diff --version 2>/dev/null >/dev/null
+if [ ! $? ]; then
+ echo "==== You must have the 'diff' program installed for this script ===="
+ exit 1
+fi
+
+# check for 'wget' program
+wget --version 2>/dev/null >/dev/null
+if [ ! $? ]; then
+ echo "==== You must have the 'wget' program installed for this script ===="
+ exit 1
+fi
+
+# make sure we're in the top-level directory
+if [ ! -d ./po ]; then
+ echo "==== No ./po directory in the current working directory ===="
+ exit 1
+fi
+
+# make sure a package argument was passed to us
+if [ -z "$1" ]; then
+ echo "Usage: $0 PACKAGE, e.g. $0 gst-plugins-good"
+ exit 1
+fi
+
+if test "$1" != "gstreamer" -a \
+ "$1" != "gst-plugins-base" -a \
+ "$1" != "gst-plugins-good" -a \
+ "$1" != "gst-plugins-ugly" -a \
+ "$1" != "gst-plugins-bad"; then
+ echo "Unexpected package '$1' ?!"
+ exit 1
+fi
+
+PACKAGE="$1"
+
+DOMAINS_TO_ADD=""
+DOMAINS_UPDATED=""
+DOMAINS_NOT_IN_LINGUAS=""
+
+echo "Downloading latest translation files for package $PACKAGE ..."
+echo
+
+for d in $DOMAINS
+do
+ PACKAGE_PO_URL_BASE="http://translationproject.org/latest/$PACKAGE"
+ PO_URL="$PACKAGE_PO_URL_BASE/$d.po"
+ PO_FILENAME="$PACKAGE.$d.po"
+ if wget -q -nc -O $PO_FILENAME $PO_URL; then
+ # we want all .po files in UTF-8 format really, so convert if needed..
+ CHARSET=`grep Content-Type $PO_FILENAME | sed -e 's/.*charset=\(.*\)\\\\n.*/\1/'`
+ if test "x$CHARSET" != "xUTF-8" -a "x$CHARSET" != "xutf-8"; then
+ # note: things like the bugs address will be added back by make update-po
+ if msguniq $PO_FILENAME --no-location \
+ --output-file=$PO_FILENAME.utf8 \
+ --to-code=UTF-8; then
+ mv $PO_FILENAME.utf8 $PO_FILENAME
+ else
+ echo "**** $d: conversion from $CHARSET to UTF-8 failed ****"
+ fi
+ fi
+ if [ -f "po/$d.po" ]; then
+ # ./po/foo.po exists, so let's check if ours matches the latest from the
+ # translation project website
+ REVDATE_NEW=`grep PO-Revision-Date $PO_FILENAME`;
+ REVDATE_OLD=`grep PO-Revision-Date po/$d.po`;
+ CHARSET_OLD=`grep Content-Type po/$d.po | sed -e 's/.*charset=\(.*\)\\\\n.*/\1/'`
+ if test "x$REVDATE_NEW" = "x$REVDATE_OLD" -a "x$CHARSET_OLD" = "xUTF-8"; then
+ # note: source code line markers will be removed later by make upload-po
+ echo "$d.po: up-to-date"
+ rm -f $PO_FILENAME
+ else
+ mv $PO_FILENAME "po/$d.po"
+ if test "x$CHARSET_OLD" != "xUTF-8" -a "x$CHARSET_OLD" != "xutf-8"; then
+ echo "$d.po: update (and charset converted from $CHARSET_OLD to UTF-8)"
+ else
+ echo "$d.po: updated"
+ fi
+ DOMAINS_UPDATED="$DOMAINS_UPDATED $d"
+ fi
+ # make sure domain is listed in LINGUAS
+ if ! grep $d "po/LINGUAS" >/dev/null 2>/dev/null; then
+ DOMAINS_NOT_IN_LINGUAS="$DOMAINS_NOT_IN_LINGUAS $d"
+ fi
+ else
+ # ./po/foo.po doesn't exist, but foo.po exists on the translation project
+ # website, so it's probably a new translation
+ echo "$d.po: new language"
+ mv $PO_FILENAME "po/$d.po"
+ DOMAINS_UPDATED="$DOMAINS_UPDATED $d"
+ DOMAINS_TO_ADD="$DOMAINS_TO_ADD $d"
+ fi
+ else
+ rm -f $PO_FILENAME
+ echo "$d.po: failure (does probably not exist)"
+ fi
+done
+
+if [ -n "$DOMAINS_UPDATED" ]; then
+ echo "===================================================================="
+ echo
+ echo "Language domains updated :$DOMAINS_UPDATED"
+ echo "Language domains to git add :$DOMAINS_TO_ADD"
+ echo
+ echo "Source: http://translationproject.org/latest/$PACKAGE/"
+ echo
+ if [ -n "$DOMAINS_TO_ADD" ]; then
+ CMD_STRING="git add"
+ for d in $DOMAINS_TO_ADD; do
+ CMD_STRING="$CMD_STRING po/$d.po"
+ done
+ echo "Please run"
+ echo
+ echo " $CMD_STRING"
+ echo
+ echo "now and add the following domains to the po/LINGUAS file:"
+ echo
+ echo " $DOMAINS_TO_ADD"
+ echo
+ echo
+ fi
+ echo "===================================================================="
+fi
+
+if [ -n "$DOMAINS_NOT_IN_LINGUAS" ]; then
+ echo
+ echo "Existing domains missing from the po/LINGUAS file:"
+ echo
+ echo " $DOMAINS_NOT_IN_LINGUAS"
+ echo
+ echo
+fi
+
+
--- /dev/null
+#!/bin/sh
+# Shell script to extract the date given a release version and a .doap file
+
+if test "x$1" = "x" -o "x$2" = "x" -o ! -s "$2"; then
+ echo "Usage: $0 RELEASE-VERSION-NUMBER DOAP-FILE" >&2;
+ exit 1
+fi
+
+if ! grep '<Project' "$2" >/dev/null ; then
+ echo "$2 does not look lika a .doap file" >&2;
+ exit 1
+fi
+
+if ! grep "$1" "$2" >/dev/null ; then
+ echo "$2 contains no reference to a version $1" >&2;
+ exit 1
+fi
+
+awk 'BEGIN {x=0}
+{
+if ( $0 ~ /<release>/ ) {x=1; chunk=""}
+if (x==1) {
+ if ($0 ~ /<revision>/) { chunk = chunk $0 }
+ if ($0 ~ /<created>/) { chunk = chunk $0 }
+}
+if ($0 ~ /<\/release>/) {x=0; print chunk}
+}' < "$2" | \
+\
+grep '<revision>'"$1"'</revision>' | \
+\
+sed -e 's/^.*<created>//' -e 's/<\/created>.*$//'
+
--- /dev/null
+--- po/Makefile.in.in.orig 2006-01-07 12:03:45.000000000 +0100
++++ po/Makefile.in.in 2006-01-07 12:04:23.000000000 +0100
+@@ -11,6 +11,9 @@
+ PACKAGE = @PACKAGE@
+ VERSION = @VERSION@
+
++# thomas: add GETTEXT_PACKAGE substitution as used in Makevars
++GETTEXT_PACKAGE = @GETTEXT_PACKAGE@
++
+ SHELL = /bin/sh
+ @SET_MAKE@
+
--- /dev/null
+# these are the variables your Makefile.am should set
+# the example is based on the colorbalance interface
+
+#glib_enum_headers=$(colorbalance_headers)
+#glib_enum_define=GST_COLOR_BALANCE
+#glib_enum_prefix=gst_color_balance
+
+enum_headers=$(foreach h,$(glib_enum_headers),\n\#include \"$(h)\")
+
+# these are all the rules generating the relevant files
+%-marshal.h: %-marshal.list
+ $(AM_V_GEN)glib-genmarshal --header --prefix=$(glib_enum_prefix)_marshal $^ > $*-marshal.h.tmp && \
+ mv $*-marshal.h.tmp $*-marshal.h
+
+%-marshal.c: %-marshal.list
+ $(AM_V_GEN)echo "#include \"$*-marshal.h\"" >> $*-marshal.c.tmp && \
+ glib-genmarshal --body --prefix=$(glib_enum_prefix)_marshal $^ >> $*-marshal.c.tmp && \
+ mv $*-marshal.c.tmp $*-marshal.c
+
+%-enumtypes.h: $(glib_enum_headers)
+ $(AM_V_GEN)glib-mkenums \
+ --fhead "#ifndef __$(glib_enum_define)_ENUM_TYPES_H__\n#define __$(glib_enum_define)_ENUM_TYPES_H__\n\n#include <glib-object.h>\n\nG_BEGIN_DECLS\n" \
+ --fprod "\n/* enumerations from \"@filename@\" */\n" \
+ --vhead "GType @enum_name@_get_type (void);\n#define GST_TYPE_@ENUMSHORT@ (@enum_name@_get_type())\n" \
+ --ftail "G_END_DECLS\n\n#endif /* __$(glib_enum_define)_ENUM_TYPES_H__ */" \
+ $^ > $@
+
+%-enumtypes.c: $(glib_enum_headers)
+ @if test "x$(glib_enum_headers)" = "x"; then echo "ERROR: glib_enum_headers is empty, please fix Makefile"; exit 1; fi
+ $(AM_V_GEN)glib-mkenums \
+ --fhead "#include \"$*-enumtypes.h\"\n$(enum_headers)" \
+ --fprod "\n/* enumerations from \"@filename@\" */" \
+ --vhead "GType\n@enum_name@_get_type (void)\n{\n static volatile gsize g_define_type_id__volatile = 0;\n if (g_once_init_enter (&g_define_type_id__volatile)) {\n static const G@Type@Value values[] = {" \
+ --vprod " { @VALUENAME@, \"@VALUENAME@\", \"@valuenick@\" }," \
+ --vtail " { 0, NULL, NULL }\n };\n GType g_define_type_id = g_@type@_register_static (\"@EnumName@\", values);\n g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);\n }\n return g_define_type_id__volatile;\n}\n" \
+ $^ > $@
+
+# a hack rule to make sure .Plo files exist because they get include'd
+# from Makefile's
+.deps/%-marshal.Plo:
+ @touch $@
+
+.deps/%-enumtypes.Plo:
+ @touch $@
--- /dev/null
+# a silly hack that generates autoregen.sh but it's handy
+# Remove the old autoregen.sh first to create a new file,
+# as the current one may be being read by the shell executing
+# this script.
+if [ -f "autoregen.sh" ]; then
+ rm autoregen.sh
+fi
+echo "#!/bin/sh" > autoregen.sh
+echo "./autogen.sh $@ \$@" >> autoregen.sh
+chmod +x autoregen.sh
+
+# helper functions for autogen.sh
+
+debug ()
+# print out a debug message if DEBUG is a defined variable
+{
+ if test ! -z "$DEBUG"
+ then
+ echo "DEBUG: $1"
+ fi
+}
+
+version_get ()
+# based on the command's version output, set variables
+# _MAJOR, _MINOR, _MICRO, _VERSION, using the given prefix as variable prefix
+#
+# arg 1: command binary name
+# arg 2: (uppercased) variable name prefix
+{
+ COMMAND=$1
+ VARPREFIX=`echo $2 | tr .,- _`
+ local ${VARPREFIX}_VERSION
+
+ # strip everything that's not a digit, then use cut to get the first field
+ pkg_version=`$COMMAND --version|head -n 1|sed 's/^.*)[^0-9]*//'|cut -d' ' -f1`
+ debug "pkg_version $pkg_version"
+ # remove any non-digit characters from the version numbers to permit numeric
+ # comparison
+ pkg_major=`echo $pkg_version | cut -d. -f1 | sed s/[a-zA-Z\-].*//g`
+ pkg_minor=`echo $pkg_version | cut -d. -f2 | sed s/[a-zA-Z\-].*//g`
+ pkg_micro=`echo $pkg_version | cut -d. -f3 | sed s/[a-zA-Z\-].*//g`
+ test -z "$pkg_major" && pkg_major=0
+ test -z "$pkg_minor" && pkg_minor=0
+ test -z "$pkg_micro" && pkg_micro=0
+ debug "found major $pkg_major minor $pkg_minor micro $pkg_micro"
+ eval ${VARPREFIX}_MAJOR=$pkg_major
+ eval ${VARPREFIX}_MINOR=$pkg_minor
+ eval ${VARPREFIX}_MICRO=$pkg_micro
+ eval ${VARPREFIX}_VERSION=$pkg_version
+}
+
+version_compare ()
+# Checks whether the version of VARPREFIX is equal to or
+# newer than the requested version
+# arg1: VARPREFIX
+# arg2: MAJOR
+# arg3: MINOR
+# arg4: MICRO
+{
+ VARPREFIX=`echo $1 | tr .,- _`
+ MAJOR=$2
+ MINOR=$3
+ MICRO=$4
+
+ eval pkg_major=\$${VARPREFIX}_MAJOR;
+ eval pkg_minor=\$${VARPREFIX}_MINOR;
+ eval pkg_micro=\$${VARPREFIX}_MICRO;
+
+ #start checking the version
+ debug "version_compare: $VARPREFIX against $MAJOR.$MINOR.$MICRO"
+
+ # reset check
+ WRONG=
+
+ if [ ! "$pkg_major" -gt "$MAJOR" ]; then
+ debug "major: $pkg_major <= $MAJOR"
+ if [ "$pkg_major" -lt "$MAJOR" ]; then
+ debug "major: $pkg_major < $MAJOR"
+ WRONG=1
+ elif [ ! "$pkg_minor" -gt "$MINOR" ]; then
+ debug "minor: $pkg_minor <= $MINOR"
+ if [ "$pkg_minor" -lt "$MINOR" ]; then
+ debug "minor: $pkg_minor < $MINOR"
+ WRONG=1
+ elif [ "$pkg_micro" -lt "$MICRO" ]; then
+ debug "micro: $pkg_micro < $MICRO"
+ WRONG=1
+ fi
+ fi
+ fi
+ if test ! -z "$WRONG"; then
+ debug "version_compare: $VARPREFIX older than $MAJOR.$MINOR.$MICRO"
+ return 1
+ fi
+ debug "version_compare: $VARPREFIX equal to/newer than $MAJOR.$MINOR.$MICRO"
+ return 0
+}
+
+
+version_check ()
+# check the version of a package
+# first argument : package name (executable)
+# second argument : optional path where to look for it instead
+# third argument : source download url
+# rest of arguments : major, minor, micro version
+# all consecutive ones : suggestions for binaries to use
+# (if not specified in second argument)
+{
+ PACKAGE=$1
+ PKG_PATH=$2
+ URL=$3
+ MAJOR=$4
+ MINOR=$5
+ MICRO=$6
+
+ # for backwards compatibility, we let PKG_PATH=PACKAGE when PKG_PATH null
+ if test -z "$PKG_PATH"; then PKG_PATH=$PACKAGE; fi
+ debug "major $MAJOR minor $MINOR micro $MICRO"
+ VERSION=$MAJOR
+ if test ! -z "$MINOR"; then VERSION=$VERSION.$MINOR; else MINOR=0; fi
+ if test ! -z "$MICRO"; then VERSION=$VERSION.$MICRO; else MICRO=0; fi
+
+ debug "major $MAJOR minor $MINOR micro $MICRO"
+
+ for SUGGESTION in $PKG_PATH; do
+ COMMAND="$SUGGESTION"
+
+ # don't check if asked not to
+ test -z "$NOCHECK" && {
+ printf " checking for $COMMAND >= $VERSION ... "
+ } || {
+ # we set a var with the same name as the package, but stripped of
+ # unwanted chars
+ VAR=`echo $PACKAGE | sed 's/-//g'`
+ debug "setting $VAR"
+ eval $VAR="$COMMAND"
+ return 0
+ }
+
+ which $COMMAND > /dev/null 2>&1
+ if test $? -eq 1;
+ then
+ debug "$COMMAND not found"
+ continue
+ fi
+
+ VARPREFIX=`echo $COMMAND | sed 's/-//g' | tr [:lower:] [:upper:]`
+ version_get $COMMAND $VARPREFIX
+
+ version_compare $VARPREFIX $MAJOR $MINOR $MICRO
+ if test $? -ne 0; then
+ echo "found $pkg_version, not ok !"
+ continue
+ else
+ echo "found $pkg_version, ok."
+ # we set a var with the same name as the package, but stripped of
+ # unwanted chars
+ VAR=`echo $PACKAGE | sed 's/-//g'`
+ debug "setting $VAR"
+ eval $VAR="$COMMAND"
+ return 0
+ fi
+ done
+
+ echo "$PACKAGE not found !"
+ echo "You must have $PACKAGE installed to compile $package."
+ echo "Download the appropriate package for your distribution,"
+ echo "or get the source tarball at $URL"
+ return 1;
+}
+
+aclocal_check ()
+{
+ # normally aclocal is part of automake
+ # so we expect it to be in the same place as automake
+ # so if a different automake is supplied, we need to adapt as well
+ # so how's about replacing automake with aclocal in the set var,
+ # and saving that in $aclocal ?
+ # note, this will fail if the actual automake isn't called automake*
+ # or if part of the path before it contains it
+ if [ -z "$automake" ]; then
+ echo "Error: no automake variable set !"
+ return 1
+ else
+ aclocal=`echo $automake | sed s/automake/aclocal/`
+ debug "aclocal: $aclocal"
+ if [ "$aclocal" != "aclocal" ];
+ then
+ CONFIGURE_DEF_OPT="$CONFIGURE_DEF_OPT --with-aclocal=$aclocal"
+ fi
+ if [ ! -x `which $aclocal` ]; then
+ echo "Error: cannot execute $aclocal !"
+ return 1
+ fi
+ fi
+}
+
+autoheader_check ()
+{
+ # same here - autoheader is part of autoconf
+ # use the same voodoo
+ if [ -z "$autoconf" ]; then
+ echo "Error: no autoconf variable set !"
+ return 1
+ else
+ autoheader=`echo $autoconf | sed s/autoconf/autoheader/`
+ debug "autoheader: $autoheader"
+ if [ "$autoheader" != "autoheader" ];
+ then
+ CONFIGURE_DEF_OPT="$CONFIGURE_DEF_OPT --with-autoheader=$autoheader"
+ fi
+ if [ ! -x `which $autoheader` ]; then
+ echo "Error: cannot execute $autoheader !"
+ return 1
+ fi
+ fi
+
+}
+
+die_check ()
+{
+ # call with $DIE
+ # if set to 1, we need to print something helpful then die
+ DIE=$1
+ if test "x$DIE" = "x1";
+ then
+ echo
+ echo "- Please get the right tools before proceeding."
+ echo "- Alternatively, if you're sure we're wrong, run with --nocheck."
+ exit 1
+ fi
+}
+
+autogen_options ()
+{
+ if test "x$1" = "x"; then
+ return 0
+ fi
+
+ while test "x$1" != "x" ; do
+ optarg=`expr "x$1" : 'x[^=]*=\(.*\)'`
+ case "$1" in
+ --noconfigure)
+ NOCONFIGURE=defined
+ AUTOGEN_EXT_OPT="$AUTOGEN_EXT_OPT --noconfigure"
+ echo "+ configure run disabled"
+ shift
+ ;;
+ --nocheck)
+ AUTOGEN_EXT_OPT="$AUTOGEN_EXT_OPT --nocheck"
+ NOCHECK=defined
+ echo "+ autotools version check disabled"
+ shift
+ ;;
+ -d|--debug)
+ DEBUG=defined
+ AUTOGEN_EXT_OPT="$AUTOGEN_EXT_OPT --debug"
+ echo "+ debug output enabled"
+ shift
+ ;;
+ -h|--help)
+ echo "autogen.sh (autogen options) -- (configure options)"
+ echo "autogen.sh help options: "
+ echo " --noconfigure don't run the configure script"
+ echo " --nocheck don't do version checks"
+ echo " --debug debug the autogen process"
+ echo
+ echo " --with-autoconf PATH use autoconf in PATH"
+ echo " --with-automake PATH use automake in PATH"
+ echo
+ echo "Any argument either not in the above list or after a '--' will be "
+ echo "passed to ./configure."
+ exit 1
+ ;;
+ --with-automake=*)
+ AUTOMAKE=$optarg
+ echo "+ using alternate automake in $optarg"
+ CONFIGURE_DEF_OPT="$CONFIGURE_DEF_OPT --with-automake=$AUTOMAKE"
+ shift
+ ;;
+ --with-autoconf=*)
+ AUTOCONF=$optarg
+ echo "+ using alternate autoconf in $optarg"
+ CONFIGURE_DEF_OPT="$CONFIGURE_DEF_OPT --with-autoconf=$AUTOCONF"
+ shift
+ ;;
+ --) shift ; break ;;
+ *)
+ echo "+ passing argument $1 to configure"
+ CONFIGURE_EXT_OPT="$CONFIGURE_EXT_OPT $1"
+ shift
+ ;;
+ esac
+ done
+
+ for arg do CONFIGURE_EXT_OPT="$CONFIGURE_EXT_OPT $arg"; done
+ if test ! -z "$CONFIGURE_EXT_OPT"
+ then
+ echo "+ options passed to configure: $CONFIGURE_EXT_OPT"
+ fi
+}
+
+toplevel_check ()
+{
+ srcfile=$1
+ test -f $srcfile || {
+ echo "You must run this script in the top-level $package directory"
+ exit 1
+ }
+}
+
+tool_run ()
+{
+ tool=$1
+ options=$2
+ run_if_fail=$3
+ echo "+ running $tool $options..."
+ $tool $options || {
+ echo
+ echo $tool failed
+ eval $run_if_fail
+ exit 1
+ }
+}
+
+install_git_hooks ()
+{
+ if test -d .git; then
+ # install pre-commit hook for doing clean commits
+ for hook in pre-commit; do
+ if test ! \( -x .git/hooks/$hook -a -L .git/hooks/$hook \); then
+ echo "+ Installing git $hook hook"
+ rm -f .git/hooks/$hook
+ ln -s ../../common/hooks/$hook.hook .git/hooks/$hook || {
+ # if we couldn't create a symbolic link, try doing a plain cp
+ if cp common/hooks/pre-commit.hook .git/hooks/pre-commit; then
+ chmod +x .git/hooks/pre-commit;
+ else
+ echo "********** Couldn't install git $hook hook **********";
+ fi
+ }
+ fi
+ done
+ fi
+}
--- /dev/null
+#!/bin/sh
+#
+# Check that the code follows a consistant code style
+#
+
+# Check for existence of indent, and error out if not present.
+# On some *bsd systems the binary seems to be called gnunindent,
+# so check for that first.
+
+version=`gnuindent --version 2>/dev/null`
+if test "x$version" = "x"; then
+ version=`indent --version 2>/dev/null`
+ if test "x$version" = "x"; then
+ echo "GStreamer git pre-commit hook:"
+ echo "Did not find GNU indent, please install it before continuing."
+ exit 1
+ fi
+ INDENT=indent
+else
+ INDENT=gnuindent
+fi
+
+case `$INDENT --version` in
+ GNU*)
+ ;;
+ default)
+ echo "GStreamer git pre-commit hook:"
+ echo "Did not find GNU indent, please install it before continuing."
+ echo "(Found $INDENT, but it doesn't seem to be GNU indent)"
+ exit 1
+ ;;
+esac
+
+INDENT_PARAMETERS="--braces-on-if-line \
+ --case-brace-indentation0 \
+ --case-indentation2 \
+ --braces-after-struct-decl-line \
+ --line-length80 \
+ --no-tabs \
+ --cuddle-else \
+ --dont-line-up-parentheses \
+ --continuation-indentation4 \
+ --honour-newlines \
+ --tab-size8 \
+ --indent-level2 \
+ --leave-preprocessor-space"
+
+$INDENT ${INDENT_PARAMETERS} $@
+
--- /dev/null
+### this file contains suppressions for valgrind when running
+### the gstreamer unit tests
+### it might be useful for wider use as well
+
+### syscall suppressions
+
+{
+ <clone on Wim's Debian>
+ Memcheck:Param
+ clone(parent_tidptr)
+ fun:clone
+ fun:clone
+}
+
+{
+ <clone on Wim's Debian>
+ Memcheck:Param
+ clone(child_tidptr)
+ fun:clone
+ fun:clone
+}
+
+{
+ <clone on Wim's Debian>
+ Memcheck:Param
+ clone(tlsinfo)
+ fun:clone
+ fun:clone
+}
+
+### glibc suppressions
+
+{
+ <conditional jump on wim's debian 2/2/06>
+ Memcheck:Cond
+ obj:/lib/ld-2.*.so
+ fun:dl_open_worker
+ obj:/lib/ld-2.*.so
+ fun:_dl_open
+ fun:dlopen_doit
+ obj:/lib/ld-2.*.so
+ fun:_dlerror_run
+ fun:dlopen
+ fun:g_module_open
+ fun:gst_plugin_load_file
+}
+
+{
+ <Conditional jump>
+ Memcheck:Cond
+ fun:strlen
+ fun:fillin_rpath
+ fun:_dl_init_paths
+ fun:dl_main
+ fun:_dl_sysdep_start
+ fun:_dl_start
+ obj:/lib64/ld-2.*.so
+ obj:*
+ obj:*
+}
+
+{
+ <Conditional jump>
+ Memcheck:Cond
+ fun:_dl_relocate_object
+ fun:dl_main
+ fun:_dl_sysdep_start
+ fun:_dl_start
+}
+
+{
+ <insert a suppression name here>
+ Memcheck:Cond
+ fun:*
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.2.5
+}
+
+# glibc does not deallocate thread-local storage
+
+{
+ <tls>
+ Memcheck:Leak
+ fun:calloc
+ fun:_dl_allocate_tls
+ fun:pthread_create@@*
+}
+
+{
+ <tls>
+ Memcheck:Leak
+ fun:calloc
+ fun:allocate_dtv
+ fun:_dl_allocate_tls
+}
+
+# I get an extra stack entry on x86/dapper
+{
+ <tls>
+ Memcheck:Leak
+ fun:calloc
+ obj:/lib/ld-2.3.*.so
+ fun:_dl_allocate_tls
+ fun:pthread_create@@*
+}
+
+
+{
+ <pthread strstr>
+ Memcheck:Cond
+ fun:strstr
+ fun:__pthread_initialize_minimal
+ obj:/lib/libpthread-*.so
+ obj:/lib/libpthread-*.so
+ fun:call_init
+ fun:_dl_init
+ obj:/lib/ld-*.so
+}
+
+# a thread-related free problem in glibc from Edgard
+{
+ __libc_freeres_rw_acess
+ Memcheck:Addr4
+ obj:*
+ obj:*
+ obj:*
+ obj:*
+ obj:*
+ fun:__libc_freeres
+}
+
+{
+ <a conditional jump on wim's debian>
+ Memcheck:Cond
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+}
+
+# g_module_open-related problems
+{
+ <started showing up on fc4-quick>
+ Memcheck:Addr2
+ fun:memcpy
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+ fun:gst_plugin_load_file
+ fun:gst_registry_scan_path_level
+ fun:gst_registry_scan_path_level
+ fun:gst_registry_scan_path_level
+ fun:init_post
+ fun:g_option_context_parse
+ fun:gst_init_check
+ fun:gst_init
+ fun:gst_check_init
+ fun:main
+}
+
+{
+ <started showing up on fc4-quick>
+ Memcheck:Addr4
+ fun:memcpy
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+ fun:gst_plugin_load_file
+ fun:gst_registry_scan_path_level
+ fun:gst_registry_scan_path_level
+ fun:gst_registry_scan_path_level
+ fun:init_post
+ fun:g_option_context_parse
+ fun:gst_init_check
+ fun:gst_init
+ fun:gst_check_init
+ fun:main
+}
+
+{
+ <g_module_open on wim's debian>
+ Memcheck:Cond
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ fun:do_sym
+ fun:_dl_sym
+ fun:dlsym_doit
+ obj:/lib/ld-2.3.*.so
+ fun:_dlerror_run
+ fun:dlsym
+ fun:g_module_symbol
+ fun:g_module_open
+ fun:gst_plugin_load_file
+}
+
+{
+ <g_module_open on wim's debian>
+ Memcheck:Cond
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ fun:dl_open_worker
+ obj:/lib/ld-2.3.*.so
+ fun:_dl_open
+ fun:dlopen_doit
+ obj:/lib/ld-2.3.*.so
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+ fun:gst_plugin_load_file
+}
+{
+ <g_module_open on wim's debian>
+ Memcheck:Cond
+ obj:/lib/ld-2.3.*.so
+ fun:dl_open_worker
+ obj:/lib/ld-2.3.*.so
+ fun:_dl_open
+ fun:dlopen_doit
+ obj:/lib/ld-2.3.*.so
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+ fun:gst_plugin_load_file
+ fun:gst_plugin_load_by_name
+ fun:gst_plugin_feature_load
+}
+
+{
+ <leak on wim's debian in g_module_open>
+ Memcheck:Leak
+ fun:malloc
+ obj:/lib/ld-2.3.*.so
+ fun:dl_open_worker
+ obj:/lib/ld-2.3.*.so
+ fun:_dl_open
+ fun:dlopen_doit
+ obj:/lib/ld-2.3.*.so
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+ fun:gst_plugin_load_file
+ fun:gst_plugin_load_by_name
+}
+
+{
+ <invalid read on wim's debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ fun:dl_open_worker
+ obj:/lib/ld-2.3.*.so
+ fun:_dl_open
+ fun:dlopen_doit
+ obj:/lib/ld-2.3.*.so
+}
+
+{
+ <invalid read on wim's debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ fun:dl_open_worker
+ obj:/lib/ld-2.3.*.so
+ fun:_dl_open
+ fun:dlopen_doit
+ obj:/lib/ld-2.3.*.so
+ fun:_dlerror_run
+}
+
+{
+ <invalid read on wim's debian - 2006-02-02>
+ Memcheck:Addr4
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ fun:dl_open_worker
+ obj:/lib/ld-2.3.*.so
+ fun:_dl_open
+ fun:dlopen_doit
+ obj:/lib/ld-2.3.*.so
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+}
+
+{
+ <invalid read on wim's debian - 2006-02-02>
+ Memcheck:Addr4
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ fun:dl_open_worker
+ obj:/lib/ld-2.3.*.so
+ fun:_dl_open
+ fun:dlopen_doit
+ obj:/lib/ld-2.3.*.so
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+}
+
+{
+ <invalid read on wim's debian - 2006-02-02>
+ Memcheck:Addr4
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ fun:do_sym
+ fun:_dl_sym
+ fun:dlsym_doit
+ obj:/lib/ld-2.3.*.so
+ fun:_dlerror_run
+ fun:dlsym
+ fun:g_module_symbol
+ fun:g_module_open
+}
+
+{
+ <futex on Andy's 64-bit ubuntu>
+ Memcheck:Param
+ futex(uaddr2)
+ fun:pthread_once
+ obj:/lib/libc-2.3.*.so
+ obj:/lib/libc-2.3.*.so
+ fun:mbsnrtowcs
+ fun:vfprintf
+ fun:vsprintf
+ fun:sprintf
+ obj:/lib/libc-2.3.*.so
+ fun:tmpfile
+ fun:setup_pipe
+ fun:setup_messaging_with_key
+ fun:setup_messaging
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ fun:_dl_sym
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlsym
+ fun:g_module_symbol
+ fun:g_module_open
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ fun:iconv_open
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ obj:/lib/i686/cmov/libc-2.7.so
+ fun:iconv_open
+}
+
+{
+ <suppression for glibc 2.7 on Ubunty Hardy 64-bit>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+ obj:/lib/libc-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/libc-2.7.so
+ fun:iconv_open
+}
+
+{
+ <suppression for glibc 2.7 on Ubunty Hardy 64-bit>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+ obj:/lib/libc-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/libc-2.7.so
+ fun:iconv_open
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on Ubunty Hardy 64-bit>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Cond
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Cond
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Cond
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Cond
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Cond
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Cond
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+{
+ <suppression for glibc 2.7 on debian>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/i686/cmov/libdl-2.7.so
+ fun:dlopen
+}
+
+# suppression for a glibc bug:
+# http://valgrind.org/docs/manual/faq.html#faq.exit_errors>
+{
+ <Workaround for a glibc bug>
+ Memcheck:Free
+ fun:free
+ obj:*libc-*.so
+ fun:__libc_freeres
+ fun:*
+ fun:_Exit
+}
+
+# same as above, just so it works for tpm on gutsy/x86-64
+{
+ <workaround glibc bug on gutsy x86-64>
+ Memcheck:Free
+ fun:free
+ fun:free_mem
+ fun:__libc_freeres
+}
+
+# valgrind doesn't allow me to specify a suppression for Addr1, Addr2, Addr4
+# as Addr*, so 3 copies for that; and then 2 of each for that pesky memcpy
+{
+ <Invalid read of size 1, 2, 4 on thomas's FC4>
+ Memcheck:Addr1
+ fun:_dl_signal_error
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+}
+
+{
+ <Invalid read of size 1, 2, 4 on thomas's FC4>
+ Memcheck:Addr2
+ fun:_dl_signal_error
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+}
+{
+ <Invalid read of size 1, 2, 4 on thomas's FC4>
+ Memcheck:Addr4
+ fun:_dl_signal_error
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+}
+
+{
+ <Invalid read of size 1, 2, 4 on thomas's FC4>
+ Memcheck:Addr1
+ fun:memcpy
+ fun:_dl_signal_error
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+}
+
+{
+ <Invalid read of size 1, 2, 4 on thomas's FC4>
+ Memcheck:Addr2
+ fun:memcpy
+ fun:_dl_signal_error
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+}
+{
+ <Invalid read of size 1, 2, 4 on thomas's FC4>
+ Memcheck:Addr4
+ fun:memcpy
+ fun:_dl_signal_error
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+ fun:g_module_open
+}
+
+{
+ <Addr8 on Andy's AMD64 ubuntu in dl_open>
+ Memcheck:Addr8
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/libc-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ fun:_dl_open
+ obj:/lib/libdl-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+}
+
+{
+ <Conditional jump on Andy's AMD64 ubuntu>
+ Memcheck:Cond
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/libc-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ fun:_dl_open
+ obj:/lib/libdl-2.3.*.so
+ obj:/lib/ld-2.3.*.so
+ obj:/lib/libdl-2.3.*.so
+ fun:dlopen
+ fun:g_module_open
+ fun:gst_plugin_load_file
+ fun:gst_plugin_load_by_name
+ fun:gst_plugin_feature_load
+}
+
+{
+ <Mike's x86 dapper>
+ Memcheck:Addr4
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/tls/i686/cmov/libc-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ fun:_dl_open
+ obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+ fun:dlopen
+}
+
+{
+ <Mike's x86 dapper>
+ Memcheck:Cond
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/tls/i686/cmov/libc-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ fun:_dl_open
+ obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+ fun:dlopen
+}
+
+{
+ <Another dapper one>
+ Memcheck:Cond
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/tls/i686/cmov/libc-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ fun:_dl_open
+ obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+ fun:dlopen
+}
+
+### glib suppressions
+{
+ <g_parse_debug_string>
+ Memcheck:Cond
+ fun:g_parse_debug_string
+ obj:/usr/lib*/libglib-2.0.so.*
+ fun:g_slice_alloc
+ fun:g_slice_alloc0
+}
+
+{
+ <g_type_init leaks>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:g_type_init*
+ fun:init_pre*
+}
+
+{
+ <g_type_register_fundamental leaks>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:g_type_register_fundamental
+}
+
+{
+ <glib 2.21 static type data>
+ Memcheck:Leak
+ fun:malloc
+ fun:realloc
+ fun:g_realloc
+ fun:type_node_any_new_W
+}
+
+{
+ <glib 2.21 static type data>
+ Memcheck:Leak
+ fun:realloc
+ fun:g_realloc
+ fun:type_node_any_new_W
+}
+
+{
+ <glib 2.21 static type data>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:g_type_class_ref
+}
+
+{
+ <glib 2.21 static type data>
+ Memcheck:Leak
+ fun:malloc
+ fun:realloc
+ fun:g_realloc
+ fun:type_add_flags_W
+}
+
+{
+ <glib 2.21 static type data>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:type_add_flags_W
+}
+
+#pthread memleaks
+
+{
+ Thread creation leak
+ Memcheck:Leak
+ fun:calloc
+ fun:allocate_dtv
+ fun:_dl_allocate*
+ fun:_dl_allocate*
+ fun:__pthread_initialize_minimal
+}
+
+{
+ Thread management leak
+ Memcheck:Leak
+ fun:calloc
+ fun:allocate_dtv
+ fun:_dl_allocate*
+ fun:_dl_allocate*
+ fun:__pthread_*
+}
+
+{
+ Thread management leak 2
+ Memcheck:Leak
+ fun:memalign
+ fun:_dl_allocate*
+ fun:_dl_allocate*
+ fun:__pthread_*
+}
+
+{
+ pthread_create Syscall param write(buf) points to uninitialised byte(s)
+ Memcheck:Param
+ write(buf)
+ fun:pthread_create@@GLIBC_2.2.5
+ fun:g_thread_create*
+
+}
+
+# nss_parse_* memleak (used by g_option_context_parse)
+{
+ nss_parse_* memleak
+ Memcheck:Leak
+ fun:malloc
+ fun:nss_parse_service_list
+ fun:__nss_database_lookup
+}
+
+# liboil suppressions
+{
+ <liboil cpu_fault_check_try>
+ Memcheck:Value8
+ obj:/usr/lib/liboil-0.3.so.0.1.0
+ obj:/usr/lib/liboil-0.3.so.0.1.0
+ obj:/usr/lib/liboil-0.3.so.0.1.0
+ fun:oil_cpu_fault_check_try
+ fun:oil_test_check_impl
+ fun:oil_class_optimize
+ fun:oil_optimize_all
+ fun:oil_init
+}
+
+{
+ <annoying read error inside dlopen stuff on Ubuntu Dapper x86_64>
+ Memcheck:Addr8
+ obj:/lib/ld-2.3.6.so
+}
+
+{
+ <Ubuntu Dapper x86_64>
+ Memcheck:Param
+ futex(uaddr2)
+ fun:pthread_once
+ obj:/lib/libc-2.3.6.so
+ obj:/lib/libc-2.3.6.so
+ fun:setlocale
+ fun:init_pre
+ fun:g_option_context_parse
+ fun:gst_init_check
+ fun:gst_init
+ fun:gst_check_init
+ fun:main
+}
+
+{
+ <Ubuntu Dapper x86_64 dlopen stuff again>
+ Memcheck:Cond
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ fun:_dl_open
+ obj:/lib/libdl-2.3.6.so
+ obj:/lib/ld-2.3.6.so
+ obj:/lib/libdl-2.3.6.so
+ fun:dlopen
+ fun:g_module_open
+ fun:gst_plugin_load_file
+}
+# this exists in a bunch of different variations, hence the short tail/trace
+{
+ <dlopen invalid read of size 4 suppression on tpm's Ubuntu edgy/x86>
+ Memcheck:Addr4
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+}
+{
+ <and the same for 64bit systems>
+ Memcheck:Addr8
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+}
+
+# More edgy suppressions (Mike)
+{
+ <dlopen Condition jump suppressions for Ubuntu Edgy/x86>
+ Memcheck:Cond
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ fun:dlopen_doit
+ obj:/lib/ld-2.4.so
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+}
+
+{
+ <dlopen Condition jump suppressions for Ubuntu Edgy/x86>
+ Memcheck:Cond
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ fun:dlopen_doit
+ obj:/lib/ld-2.4.so
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.1
+}
+
+{
+ <dlopen Condition jump suppressions for Ubuntu Edgy/x86>
+ Memcheck:Cond
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ fun:do_sym
+ fun:_dl_sym
+}
+
+# This one's overly general, but there's zero other information in the stack
+# trace - just these five lines!
+{
+ <dlopen Condition jump suppressions for Ubuntu Edgy/x86>
+ Memcheck:Cond
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+ obj:/lib/ld-2.4.so
+}
+
+{
+ <tls leaks on Edgy/x86>
+ Memcheck:Leak
+ fun:calloc
+ obj:/lib/ld-2.4.so
+ fun:_dl_allocate_tls
+ fun:pthread_create@@GLIBC_2.1
+}
+
+# TLS leaks for feisty/x86
+{
+ <tls leaks on Feisty/x86>
+ Memcheck:Leak
+ fun:calloc
+ fun:allocate_dtv
+ fun:_dl_allocate_tls
+ fun:pthread_create@@GLIBC_2.1
+}
+
+{
+ <libcdio 0.76 leak>
+ Memcheck:Leak
+ fun:calloc
+ obj:/usr/lib/libcdio.so.6.0.1
+ fun:cdio_open_am_linux
+ obj:/usr/lib/libcdio.so.6.0.1
+ fun:cdio_open_am
+}
+
+{
+ <Addr8 on Jan's AMD64 ubuntu Feisty in dl_open>
+ Memcheck:Addr8
+ obj:/lib/ld-2.5.so
+}
+
+{
+ <First of many Alsa errors>
+ Memcheck:Cond
+ fun:snd_pcm_direct_shm_create_or_connect
+ fun:snd_pcm_dsnoop_open
+ fun:_snd_pcm_dsnoop_open
+ obj:/*lib/libasound.so.2.0.0
+ obj:/*lib/libasound.so.2.0.0
+ fun:snd_pcm_open_slave
+ fun:_snd_pcm_plug_open
+ obj:/*lib/libasound.so.2.0.0
+ fun:snd_pcm_open_slave
+ fun:_snd_pcm_asym_open
+ obj:/*lib/libasound.so.2.0.0
+ obj:/*lib/libasound.so.2.0.0
+}
+
+{
+ <alsa error>
+ Memcheck:Cond
+ fun:snd*_pcm_hw_param_set_near
+}
+
+{
+ <alsa error>
+ Memcheck:Cond
+ ...
+ fun:snd*_pcm_hw_param_set_near
+}
+
+{
+ <alsa error>
+ Memcheck:Cond
+ obj:/*lib/libasound.so.2.0.0
+ obj:/*lib/libasound.so.2.0.0
+ fun:snd_pcm_close
+ obj:/*lib/libasound.so.2.0.0
+}
+{
+ <alsa error>
+ Memcheck:Cond
+ fun:snd_pcm_direct_shm_create_or_connect
+ fun:snd_pcm_dmix_open
+ fun:_snd_pcm_dmix_open
+ obj:/*lib/libasound.so.2.0.0
+ obj:/*lib/libasound.so.2.0.0
+ fun:snd_pcm_open_slave
+ fun:_snd_pcm_softvol_open
+ obj:/*lib/libasound.so.2.0.0
+ fun:snd_pcm_open_slave
+ fun:_snd_pcm_plug_open
+ obj:/*lib/libasound.so.2.0.0
+ fun:snd_pcm_open_slave
+ fun:_snd_pcm_asym_open
+ obj:/*lib/libasound.so.2.0.0
+ obj:/*lib/libasound.so.2.0.0
+}
+{
+ <alsa error>
+ Memcheck:Leak
+ fun:malloc
+ fun:strdup
+ fun:snd_dlobj_cache_add
+ obj:/*lib/libasound.so.2.0.0
+ fun:snd_pcm_open_slave
+ fun:snd_pcm_dsnoop_open
+ fun:_snd_pcm_dsnoop_open
+ obj:/*lib/libasound.so.2.0.0
+ obj:/*lib/libasound.so.2.0.0
+ fun:snd_pcm_open_slave
+ fun:_snd_pcm_plug_open
+ obj:/*lib/libasound.so.2.0.0
+ fun:snd_pcm_open_slave
+ fun:_snd_pcm_asym_open
+ obj:/*lib/libasound.so.2.0.0
+ obj:/*lib/libasound.so.2.0.0
+}
+# Catch about 15 variations on inserting info into an ALSA
+# internal cache
+{
+ <alsa error>
+ Memcheck:Leak
+ fun:malloc
+ ...
+ fun:snd*_dlobj_cache_add
+ obj:/*lib*/libasound.so.2.0.0
+}
+
+{
+ <alsa leak in loading configuration>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:snd_pcm_open_conf
+}
+
+{
+ <alsa leak snd_config_hook_load>
+ Memcheck:Leak
+ fun:*alloc
+ obj:/*lib*/libasound.so.2.0.0
+ ...
+ fun:snd_config_hook_load
+}
+
+{
+ <alsa leak snd_config_update_r>
+ Memcheck:Leak
+ fun:*alloc
+ obj:/*lib*/libasound.so.2.0.0
+ ...
+ fun:snd_config_update_r
+ fun:snd_config_update
+}
+{
+ <alsa leak snd_config_update_r>
+ Memcheck:Leak
+ fun:*alloc
+ fun:strdup
+ ...
+ fun:snd_config_update_r
+ fun:snd_config_update
+}
+{
+ <alsa leak snd_config_searcha_hooks>
+ Memcheck:Leak
+ fun:*alloc
+ fun:_dl_close_worker
+ ...
+ fun:snd_config_searcha_hooks
+}
+
+{
+ <nss lookup within ALSA>
+ Memcheck:Leak
+ fun:malloc
+ obj:/lib/libc*.so
+ fun:__nss_database_lookup
+ obj:*
+ obj:*
+ fun:getgrnam_r
+ fun:getgrnam
+ fun:snd_pcm_direct_parse_open_conf
+}
+
+{
+ <libxcb leak on Ubuntu Feisty>
+ Memcheck:Leak
+ fun:calloc
+ fun:_XCBInitDisplayLock
+ fun:XOpenDisplay
+}
+
+# GConf internal initialisations related to getting the default client.
+{
+ <Orbit something or other>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc_tcval
+ obj:/usr/lib/libORBit-2.so.*
+ fun:ORBit_demarshal_IOR
+ fun:ORBit_demarshal_object
+ fun:CORBA_ORB_string_to_object
+ obj:/usr/lib/libgconf-2.so.*
+ fun:gconf_get_current_lock_holder
+ fun:gconf_activate_server
+ obj:/usr/lib/libgconf-2.so.*
+ obj:/usr/lib/libgconf-2.so.*
+ fun:gconf_engine_get_default
+}
+{
+ <gconf internal leak>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc_tcval
+ obj:*
+ fun:PortableServer_POA_servant_to_reference
+ fun:*
+ fun:*
+ fun:*
+ fun:gconf_engine_get_default
+}
+{
+ <gconf internal leak>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc_tcval
+ obj:/usr/lib/libORBit-2.so.*
+ fun:ORBit_demarshal_IOR
+ fun:ORBit_demarshal_object
+ fun:CORBA_ORB_string_to_object
+ obj:/usr/lib/libgconf-2.so.*
+ fun:gconf_get_current_lock_holder
+ fun:gconf_activate_server
+ obj:/usr/lib/libgconf-2.so.*
+ obj:/usr/lib/libgconf-2.so.*
+ fun:gconf_engine_get_default
+}
+{
+ <gconf internal initialisation>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc*
+ fun:*
+ fun:ORBit_demarshal_IOR
+ fun:ORBit_demarshal_object
+ fun:ORBit_demarshal_value
+ fun:*
+ fun:ORBit_small_invoke_stub
+ fun:ConfigServer_get_default_database
+ fun:*
+ fun:gconf_engine_get_default
+}
+{
+ <gconf internal init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc*
+ fun:*
+ fun:IOP_generate_profiles
+ fun:ORBit_marshal_object
+ fun:ORBit_marshal_value
+ fun:*
+ fun:ORBit_small_invoke_stub
+ fun:ConfigServer_add_client
+ fun:*
+ fun:*
+ fun:gconf_engine_get_default
+}
+{
+ <gconf internal init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc_by_tc
+ fun:*
+ fun:PortableServer_POA_servant_to_reference
+ fun:*
+ fun:*
+ fun:*
+ fun:gconf_engine_get_default
+}
+{
+ <gconf internal init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc_by_tc
+ obj:/usr/lib/libORBit-2.so.*
+ fun:ORBit_demarshal_IOR
+ fun:ORBit_demarshal_object
+ fun:CORBA_ORB_string_to_object
+ obj:/usr/lib/libgconf-2.so.*
+ fun:gconf_get_current_lock_holder
+ fun:gconf_activate_server
+ obj:/usr/lib/libgconf-2.so.*
+ obj:/usr/lib/libgconf-2.so.*
+ fun:gconf_engine_get_default
+}
+
+{
+ <insert a suppression name here>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc*
+ fun:*
+ fun:ORBit_demarshal_IOR
+ fun:ORBit_demarshal_object
+ fun:*
+ fun:*
+ fun:gconf_activate_server
+}
+
+# Some libORBit/bonobo initialisation stuff
+{
+ <bonobo init>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:ORBit_alloc_string
+ fun:CORBA_string_dup
+ fun:Bonobo_ActivationEnvValue_set
+ fun:bonobo_activation_init_activation_env
+ fun:bonobo_activation_orb_init
+ fun:bonobo_activation_init
+}
+{
+ <bonobo init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc*
+ fun:ORBit_small_alloc*
+ obj:/usr/lib/libORBit-2.so*
+ fun:PortableServer_POA_servant_to_reference
+ obj:/usr/lib/libbonobo-2.so*
+}
+{
+ <bonobo init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc_tcval
+ fun:ORBit_small_allocbuf
+ fun:ORBit_adaptor_setup
+ obj:/usr/lib/libORBit-2.so*
+ fun:ORBit_POA_setup_root
+ fun:ORBit_init_internals
+ fun:CORBA_ORB_init
+}
+{
+ <bonobo init - more recent variant of above>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc_tcval
+ fun:ORBit_adaptor_setup
+ fun:*
+ fun:ORBit_POA_setup_root
+ fun:ORBit_init_internals
+ fun:CORBA_ORB_init
+}
+{
+ <bonobo init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc*
+ fun:ORBit_small_allocbuf
+ fun:bonobo_activation_init_activation_env
+ fun:bonobo_activation_orb_init
+ fun:bonobo_activation_init
+}
+
+# More GConf stuff from the FC5 buildbot, mostly variations on the
+# above stack traces
+{
+ <incompletely initialised ORBit buffer>
+ Memcheck:Param
+ writev(vector[...])
+ fun:writev
+ obj:/usr/lib/libORBit-2.so*
+ fun:link_connection_writev
+ fun:giop_send_buffer_write
+ obj:/usr/lib/libORBit-2.so*
+ fun:ORBit_small_invoke_stub
+ fun:ORBit_small_invoke_stub_n
+ fun:ORBit_c_stub_invoke
+ fun:ConfigServer_ping
+ fun:gconf_activate_server
+ obj:/usr/lib/libgconf-2.so*
+ obj:/usr/lib/libgconf-2.so*
+ fun:gconf_engine_get_default
+}
+{
+ <gconf init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc*
+ fun:ORBit_small_alloc*
+ obj:/usr/lib/libORBit-2.so*
+ fun:PortableServer_POA_servant_to_reference
+ obj:/usr/lib/libgconf-2.so*
+ obj:/usr/lib/libgconf-2.so*
+ obj:/usr/lib/libgconf-2.so*
+ fun:gconf_engine_get_default
+}
+{
+ <gconf init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc*
+ fun:ORBit_small_alloc
+ obj:/usr/lib/libORBit-2.so*
+ fun:ORBit_demarshal_IOR
+ fun:ORBit_demarshal_object
+ fun:CORBA_ORB_string_to_object
+ obj:/usr/lib/libgconf-2.so*
+ fun:gconf_get_current_lock_holder
+ fun:gconf_activate_server
+ obj:/usr/lib/libgconf-2.so*
+ obj:/usr/lib/libgconf-2.so*
+ fun:gconf_engine_get_default
+}
+{
+ <gconf init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc*
+ fun:ORBit_small_alloc*
+ obj:/usr/lib/libORBit-2.so*
+ fun:ORBit_demarshal_IOR
+ fun:ORBit_demarshal_object
+ fun:CORBA_ORB_string_to_object
+ obj:/usr/lib/libgconf-2.so*
+ fun:gconf_get_current_lock_holder
+ fun:gconf_activate_server
+ obj:/usr/lib/libgconf-2.so*
+ obj:/usr/lib/libgconf-2.so*
+ fun:gconf_engine_get_default
+}
+{
+ <bonobo init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc*
+ fun:ORBit_small_alloc*
+ obj:/usr/lib/libORBit-2.so*
+ fun:ORBit_demarshal_IOR
+ fun:ORBit_demarshal_object
+ fun:ORBit_demarshal_value
+ obj:/usr/lib/libORBit-2.so*
+ fun:ORBit_small_invoke_stub
+ fun:ORBit_small_invoke_stub_n
+ fun:ORBit_c_stub_invoke
+ fun:ConfigServer_get_default_database
+ obj:/usr/lib/libgconf-2.so*
+ fun:gconf_engine_get_default
+}
+{
+ <gconf init>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:ORBit_alloc*
+ fun:ORBit_small_alloc*
+ obj:/usr/lib/libORBit-2.so*
+ fun:ORBit_OAObject_object_to_objkey
+ fun:IOP_generate_profiles
+ fun:ORBit_marshal_object
+ fun:ORBit_marshal_value
+ obj:/usr/lib/libORBit-2.so*
+ fun:ORBit_small_invoke_stub
+ fun:ORBit_small_invoke_stub_n
+ fun:ORBit_c_stub_invoke
+ fun:ConfigServer_add_client
+ obj:/usr/lib/libgconf-2.so*
+ obj:/usr/lib/libgconf-2.so*
+ fun:gconf_engine_get_default
+}
+{
+ <GLib caching the home dir>
+ Memcheck:Leak
+ fun:malloc
+ obj:*libc-*.so
+ fun:__nss_database_lookup
+ obj:*
+ obj:*
+ fun:getpwnam_r
+ obj:/usr/lib*/libglib-2.0.so.*
+ fun:g_get_home_dir
+}
+{
+ <GLib caching the user name>
+ Memcheck:Leak
+ fun:malloc
+ obj:*libc-*.so
+ fun:__nss_database_lookup
+ obj:*
+ obj:*
+ fun:getpwnam_r
+ obj:/usr/lib*/libglib-2.0.so.*
+ fun:g_get_user_name
+}
+{
+ <GLib caching the tmp dir>
+ Memcheck:Leak
+ fun:malloc
+ obj:*libc-*.so
+ fun:__nss_database_lookup
+ obj:*
+ obj:*
+ fun:getpwnam_r
+ obj:/usr/lib*/libglib-2.0.so.*
+ fun:g_get_tmp_dir
+}
+
+{
+ <GLib caching the host name>
+ Memcheck:Leak
+ fun:malloc
+ obj:*libc-*.so
+ fun:__nss_database_lookup
+ obj:*
+ obj:*
+ fun:getpwnam_r
+ obj:/usr/lib*/libglib-2.0.so.0.*
+ fun:g_get_host_name
+}
+
+
+## Some Fontconfig errors.
+{
+ <First time load of a font - feisty x86_64>
+ Memcheck:Leak
+ fun:malloc
+ fun:FcPatternObjectInsertElt
+ fun:FcPatternObjectAddWithBinding
+ fun:FcPatternAppend
+ fun:FcEndElement
+ obj:/usr/lib/libexpat.so.*
+ obj:/usr/lib/libexpat.so.*
+ obj:/usr/lib/libexpat.so.*
+ obj:/usr/lib/libexpat.so.*
+ fun:XML_ParseBuffer
+ fun:FcConfigParseAndLoad
+ fun:FcConfigParseAndLoad
+ fun:FcParseInclude
+ fun:FcEndElement
+ obj:/usr/lib/libexpat.so.*
+ obj:/usr/lib/libexpat.so.*
+ obj:/usr/lib/libexpat.so.*
+ obj:/usr/lib/libexpat.so.*
+ fun:XML_ParseBuffer
+ fun:FcConfigParseAndLoad
+}
+{
+ <First time load of a font - generic>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:FcInitLoadConfig
+}
+
+# Issues with ubuntu Hardy, same crack as for previous ubuntus
+{
+ <tls leak generic ubuntu hardy x86>
+ Memcheck:Leak
+ fun:calloc
+ obj:*
+ fun:_dl_allocate_tls
+ fun:pthread_create@@*
+ obj:/usr/lib/libgthread*
+ fun:g_thread_*
+}
+
+# I've made this version generic, so that it covers future modifications
+# of library names
+{
+ <tls leak generic>
+ Memcheck:Leak
+ fun:calloc
+ obj:*
+ fun:_dl_allocate_tls
+ fun:pthread_create@@*
+ fun:g_thread_*
+}
+
+# series of invalid read of size 4 in g_module_open for ubuntu
+# hardy x86/32bit
+{
+ <invalid read of size 4 within <g_module_open>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+ fun:gst_plugin_load_*
+}
+
+{
+ <invalid read of size 4 within <g_module_open>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+ fun:gst_plugin_load_*
+}
+
+{
+ <invalid read of size 4 within <g_module_open>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+ fun:gst_plugin_load_*
+}
+
+{
+ <invalid read of size 4 within <g_module_open>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+ fun:gst_plugin_load_*
+}
+
+{
+ <invalid read of size 4 within <g_module_open>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+ fun:gst_plugin_load*
+}
+
+{
+ <invalid read of size 4 within <g_module_open>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libc-2.7.so
+ fun:_dl_sym
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libdl-2.7.so
+ fun:dlsym
+ fun:g_module_symbol
+ fun:g_module_open
+ fun:gst_plugin_load_*
+}
+
+# series of invalid read of size 8 in g_module_open for ubuntu
+# hardy x86/64bit
+{
+ <invalid read of size 8 within <g_module_open>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+}
+
+{
+ <invalid read of size 8 within <g_module_open>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+}
+
+{
+ <invalid read of size 8 within <g_module_open>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+}
+
+{
+ <invalid read of size 8 within <g_module_open>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+}
+
+{
+ <invalid read of size 8 within <g_module_open>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+}
+
+{
+ <invalid read of size 8 within <g_module_open>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ fun:dlopen
+ fun:g_module_open
+}
+
+{
+ <invalid read of size 8 within <g_module_open>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/libdl-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libdl-2.7.so
+ fun:dlsym
+ fun:g_module_symbol
+ fun:g_module_open
+}
+
+{
+ <GLib caching>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+ fun:__nss_lookup_function
+ obj:/lib/tls/i686/cmov/libc-2.7.so
+ fun:__nss_passwd_lookup
+ fun:getpwnam_r
+}
+
+{
+ <GLib caching>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+ fun:__nss_lookup_function
+ obj:/lib/tls/i686/cmov/libc-2.7.so
+ fun:__nss_passwd_lookup
+ fun:getpwnam_r
+}
+
+{
+ <GLib caching>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+ fun:__nss_lookup_function
+ obj:/lib/tls/i686/cmov/libnss_compat-2.7.so
+ fun:_nss_compat_getpwnam_r
+ fun:getpwnam_r
+}
+
+{
+ <GLib caching>
+ Memcheck:Addr4
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/tls/i686/cmov/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+ fun:__nss_lookup_function
+ obj:/lib/tls/i686/cmov/libnss_compat-2.7.so
+ fun:_nss_compat_getpwnam_r
+ fun:getpwnam_r
+}
+
+{
+ <GLib caching>
+ Memcheck:Addr8
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/ld-2.7.so
+ obj:/lib/libc-2.7.so
+ obj:/lib/ld-2.7.so
+ fun:__libc_dlopen_mode
+ fun:__nss_lookup_function
+ obj:/lib/libc-2.7.so
+ fun:getpwnam_r
+}
+
+## Leaks in ALSA (variations of leak from snd_config_load1)
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:calloc
+ fun:_snd_config_make
+ fun:_snd_config_make_add
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:calloc
+ fun:_snd_config_make
+ fun:_snd_config_make_add
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:calloc
+ fun:_snd_config_make
+ fun:_snd_config_make_add
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:calloc
+ fun:_snd_config_make
+ fun:_snd_config_make_add
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:calloc
+ fun:_snd_config_make
+ fun:_snd_config_make_add
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:calloc
+ fun:_snd_config_make
+ fun:_snd_config_make_add
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:calloc
+ fun:_snd_config_make
+ fun:_snd_config_make_add
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:malloc
+ fun:snd1_dlobj_cache_add
+ fun:snd_ctl_open_noupdate
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:malloc
+ fun:*
+ fun:snd1_dlobj_cache_add
+ fun:snd_ctl_open_noupdate
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+{
+ <Alsa leak>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:snd_config_load1
+}
+
+
+# The following are leaks of caps that need to be created dynamically
+# in the type registration of the plugin (used for pad templates).
+
+{
+ <Leak in ogmparsers>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:gst_caps_new_empty
+ fun:gst_caps_new_simple
+ fun:*
+ fun:g_type_class_ref
+ fun:gst_element_register
+}
+
+{
+ <Leak in ogmparsers>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:gst_caps_new_empty
+ fun:*
+ fun:*
+ fun:g_type_class_ref
+ fun:gst_element_register
+ fun:gst_ogm_parse_plugin_init
+ fun:plugin_init
+}
+
+{
+ <Leak in videotestsrc>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:gst_caps_new_empty
+ fun:gst_caps_copy
+ fun:gst_video_test_src_base_init
+ fun:g_type_class_ref
+ fun:gst_element_register
+}
+
+{
+ <Leak in videotestsrc>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:gst_caps_new_empty
+ fun:gst_caps_copy
+ fun:gst_video_test_src_getcaps
+ fun:gst_video_test_src_base_init
+ fun:g_type_class_ref
+ fun:gst_element_register
+}
+
+{
+ <Leak in ffmpegcolorspace>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:gst_caps_new_empty
+ fun:gst_ffmpegcsp_codectype_to_caps
+ fun:gst_ffmpegcolorspace_register
+ fun:plugin_init
+}
+
+{
+ <Leak in ffmpegocolorspace>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:gst_caps_new_empty
+ fun:gst_caps_copy
+ fun:gst_ffmpegcolorspace_register
+ fun:plugin_init
+}
+
+{
+ <Leak in gstffmpegdemux>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:gst_caps_new_empty
+ fun:gst_caps_new_any
+ fun:gst_ffmpegdemux_register
+ fun:plugin_init
+}
+
+{
+ <Leak in GstAudioFilter subclasses>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:gst_caps_new_empty
+ fun:gst_caps_copy
+ fun:gst_audio_filter_class_add_pad_templates
+}
+
+{
+ <Leak in GstAudioFilter subclasses, variant>
+ Memcheck:Leak
+ fun:realloc
+ fun:g_realloc
+ fun:g_ptr_array_maybe_expand
+ fun:g_ptr_array_add
+ fun:gst_caps_append
+ fun:gst_audio_filter_class_add_pad_templates
+}
+
+{
+ <Leak in GstAudioFilter subclasses, variant>
+ Memcheck:Leak
+ fun:malloc
+ fun:realloc
+ fun:g_realloc
+ fun:g_ptr_array_maybe_expand
+ fun:g_ptr_array_add
+ fun:gst_caps_append
+ fun:gst_audio_filter_class_add_pad_templates
+}
+
+{
+ <Leak in GstAudioFilter subclasses, variant>
+ Memcheck:Leak
+ fun:malloc
+ fun:realloc
+ fun:g_realloc
+ fun:g_ptr_array_maybe_expand
+ fun:g_ptr_array_add
+ fun:gst_caps_copy
+ fun:gst_audio_filter_class_add_pad_templates
+}
+
+{
+ <Leak in GstAudioFilter subclasses, variant2>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:g_ptr_array_sized_new
+ fun:gst_caps_new_empty
+ fun:gst_caps_copy
+ fun:gst_audio_filter_class_add_pad_templates
+}
+{
+ <Leak in GstAudioFilter subclasses, variant3>
+ Memcheck:Leak
+ fun:malloc
+ fun:realloc
+ fun:g_realloc
+ fun:g_array_maybe_expand
+ fun:g_array_sized_new
+ fun:*
+ fun:*
+ fun:*
+ fun:gst_value_init_and_copy
+ fun:gst_structure_copy
+ fun:gst_caps_copy
+ fun:gst_audio_filter_class_add_pad_templates
+}
+{
+ <Leak in GstAudioFilter subclasses, variant4>
+ Memcheck:Leak
+ fun:malloc
+ fun:realloc
+ fun:g_realloc
+ fun:g_array_maybe_expand
+ fun:g_array_sized_new
+ fun:*
+ fun:gst_structure_copy
+ fun:gst_caps_copy
+ fun:gst_audio_filter_class_add_pad_templates
+}
+{
+ <Leak in GstAudioFilter subclasses, variant5>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:g_array_sized_new
+ fun:*
+ fun:gst_structure_copy
+ fun:gst_caps_copy
+ fun:gst_audio_filter_class_add_pad_templates
+}
+
+{
+ <Leak in riff-media>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:gst_caps_new_empty
+ fun:gst_riff_create_*_template_caps
+}
+{
+ <Leak in riff-media>
+ Memcheck:Leak
+ fun:malloc
+ fun:realloc
+ fun:g_realloc
+ fun:*
+ fun:*
+ fun:*
+ fun:gst_structure_copy
+ fun:gst_caps_copy
+ fun:gst_caps_append
+ fun:gst_riff_create_*_template_caps
+}
+{
+ <Leak in riff-media>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:g_array_sized_new
+ fun:*
+ fun:gst_structure_copy
+ fun:gst_caps_copy
+ fun:gst_caps_append
+ fun:gst_riff_create_*_template_caps
+}
+
+## Leaks in pango (bilboed: gentoo unstable amd64)
+
+{
+ <Pango leak - generic>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:pango_layout_get_pixel_extents
+}
+{
+ <insert a suppression name here>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:pango_language_from_string
+ fun:pango_language_get_default
+ fun:pango_context_init
+ fun:g_type_create_instance
+ fun:g_object_constructor
+ fun:g_object_newv
+ fun:g_object_new_valist
+ fun:g_object_new
+ fun:pango_font_map_create_context
+}
+
+{
+ <PangoLanguage can never be freed>
+ Memcheck:Leak
+ fun:calloc
+ fun:g_malloc0
+ fun:pango_language_from_string
+}
+
+
+## Leak of everything allocated by gst-libav plugin init
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:gst_ffmpeg_cfg_init
+}
+
+## Leak of GIO module through gnomevfs
+
+{
+ <gio leak>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:*
+ fun:*
+ fun:g_type_create_instance
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:g_io_module_new
+ fun:g_io_modules_load_all_in_directory
+ fun:*
+ fun:get_default_vfs
+}
+
+## Conditional jump in getaddrinfo (bilboed, gentoo ~amd64, Dec 13 2008)
+{
+ <Leak of addrinfo in esd>
+ Memcheck:Cond
+ fun:gaih_inet
+ fun:getaddrinfo
+}
+
+## Dynamic pad templates in mxfmux
+{
+ <Dynamic pad templates in mxfmux>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_slice_alloc
+ fun:gst_caps_new_empty
+ fun:gst_caps_from_string
+ fun:mxf_*_init
+ fun:plugin_init
+}
+
+## We don't know if ffmpeg frees this or not and better pass a copy for safety
+{
+ <insert a suppression name here>
+ Memcheck:Leak
+ fun:malloc
+ fun:g_malloc
+ fun:g_strdup
+ fun:gst_ffmpeg_cfg_fill_context
+ fun:gst_ffmpegenc_setcaps
+ fun:gst_pad_set_caps
+}
+
+## Leak/overreads with glibc-2.10
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:do_sym
+ fun:dlsym_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlsym
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:do_sym
+ fun:dlsym_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlsym
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen*
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_relocate_object
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen*
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_check_map_versions
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen*
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen*
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_relocate_object
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen*
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_check_map_versions
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen*
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_map_object*
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen*
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_map_object*
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen*
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_check_caller
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen*
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_check_caller
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen*
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ obj:/lib*/libc-2.10.*.so
+ obj:/lib*/libc-2.10.*.so
+ fun:_vgnU_freeres
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ obj:/lib*/libc-2.10.*.so
+ obj:/lib*/libc-2.10.*.so
+ fun:_vgnU_freeres
+}
+{
+ <glibc-2.10 mysterious invalid free on exit>
+ Memcheck:Free
+ fun:free
+ obj:/lib*/libc-2.10.*.so
+ obj:/lib*/libc-2.10.*.so
+ fun:_vgnU_freeres
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_fini
+ fun:__run_exit_handlers
+ fun:exit
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_fini
+ fun:__run_exit_handlers
+ fun:exit
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_sort_fini
+ fun:_dl_fini
+ fun:__run_exit_handlers
+ fun:exit
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_sort_fini
+ fun:_dl_fini
+ fun:__run_exit_handlers
+ fun:exit
+}
+
+# glibc-2.10 dl overreads
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_fixup
+ fun:_dl_runtime_resolve
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_fixup
+ fun:_dl_runtime_resolve
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_lookup_symbol_x
+ fun:_dl_fixup
+ fun:_dl_runtime_resolve
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_lookup_symbol_x
+ fun:_dl_fixup
+ fun:_dl_runtime_resolve
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:call_init
+ fun:_dl_init
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_init
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:do_lookup_x
+ fun:_dl_lookup_symbol_x
+ fun:_dl_relocate_object
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:do_lookup_x
+ fun:_dl_lookup_symbol_x
+ fun:_dl_relocate_object
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_lookup_symbol_x
+ fun:_dl_relocate_object
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_relocate_object
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:dl_main
+ fun:_dl_sysdep_start
+ fun:_dl_start
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:dl_main
+ fun:_dl_sysdep_start
+ fun:_dl_start
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:do_lookup_x
+ fun:_dl_lookup_symbol_x
+ fun:_dl_relocate_object
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:do_lookup_x
+ fun:_dl_lookup_symbol_x
+ fun:_dl_relocate_object
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_check_map_versions
+ fun:_dl_check_all_versions
+ fun:version_check_doit
+ fun:_dl_receive_error
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_check_map_versions
+ fun:_dl_check_all_versions
+ fun:version_check_doit
+ fun:_dl_receive_error
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_check_all_versions
+ fun:version_check_doit
+ fun:_dl_receive_error
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_check_all_versions
+ fun:version_check_doit
+ fun:_dl_receive_error
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_check_map_versions
+ fun:_dl_check_all_versions
+ fun:version_check_doit
+ fun:_dl_receive_error
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:_dl_check_map_versions
+ fun:_dl_check_all_versions
+ fun:version_check_doit
+ fun:_dl_receive_error
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:init_tls
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:init_tls
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_protect_relro
+ fun:_dl_relocate_object
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:do_lookup_x
+ fun:_dl_lookup_symbol_x
+ fun:_dl_relocate_object
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_setup_hash
+ fun:_dl_map_object_from_fd
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_new_object
+ fun:_dl_map_object_from_fd
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:_dl_new_object
+ fun:_dl_map_object_from_fd
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:openaux
+ fun:_dl_catch_error
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_map_object
+ fun:openaux
+ fun:_dl_catch_error
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_map_object
+ fun:openaux
+ fun:_dl_catch_error
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:_dl_map_object
+ fun:openaux
+ fun:_dl_catch_error
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:open_path
+ fun:_dl_map_object
+ fun:openaux
+ fun:_dl_catch_error
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:open_path
+ fun:_dl_map_object
+ fun:openaux
+ fun:_dl_catch_error
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_map_object_from_fd
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_map_object_from_fd
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_new_object
+ fun:_dl_map_object_from_fd
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_new_object
+ fun:_dl_map_object_from_fd
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_new_object
+ fun:_dl_map_object_from_fd
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_name_match_p
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:*
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_name_match_p
+ fun:_dl_check_map_versions
+ fun:_dl_check_all_versions
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:*
+ fun:do_lookup_x
+ fun:_dl_lookup_symbol_x
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:do_lookup_x
+ fun:_dl_lookup_symbol_x
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:do_lookup_x
+ fun:_dl_lookup_symbol_x
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:do_lookup_x
+ fun:_dl_lookup_symbol_x
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:do_lookup_x
+ fun:_dl_lookup_symbol_x
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_name_match_p
+ fun:_dl_map_object
+ fun:dl_open_worker
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_name_match_p
+ fun:_dl_map_object
+ fun:dl_open_worker
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_name_match_p
+ fun:_dl_map_object
+ fun:dl_open_worker
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:_dl_name_match_p
+ fun:_dl_map_object
+ fun:dl_open_worker
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_lookup_symbol_x
+ fun:_dl_relocate_object
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_lookup_symbol_x
+ fun:_dl_relocate_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:*
+ fun:_dl_check_map_versions
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_check_map_versions
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:*
+ fun:_dl_check_map_versions
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:_dl_check_map_versions
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:openaux
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_name_match_p
+ fun:_dl_map_object
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_close_worker
+ fun:_dl_close
+ fun:_dl_catch_error
+ fun:dlerror_run
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_close_worker
+ fun:_dl_close
+ fun:_dl_catch_error
+ fun:dlerror_run
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:_dl_close_worker
+ fun:_dl_close
+ fun:_dl_catch_error
+ fun:dlerror_run
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_close_worker
+ fun:_dl_close
+ fun:_dl_catch_error
+ fun:dlerror_run
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:fillin_rpath
+ fun:_dl_init_paths
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:fillin_rpath
+ fun:_dl_init_paths
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:fillin_rpath
+ fun:_dl_init_paths
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:fillin_rpath
+ fun:_dl_init_paths
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_map_object
+ fun:map_doit
+ fun:_dl_catch_error
+ fun:do_preload
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_map_object
+ fun:map_doit
+ fun:_dl_catch_error
+ fun:do_preload
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Param
+ open(filename)
+ fun:open
+ fun:open_verify
+ fun:_dl_map_object
+ fun:map_doit
+ fun:_dl_catch_error
+ fun:do_preload
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Param
+ stat(file_name)
+ fun:_xstat
+ fun:open_path
+ fun:_dl_map_object
+ fun:openaux
+ fun:_dl_catch_error
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_catch_error
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:*
+ fun:_dl_map_object_deps
+ fun:dl_main
+}
+
+# glibc-2.10 tls issues
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:init_tls
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:init_tls
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:*
+ fun:init_tls
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:*
+ fun:init_tls
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:_dl_allocate_tls_init
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:_dl_allocate_tls_init
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:*
+ fun:_dl_allocate_tls_init
+ fun:dl_main
+}
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Value8
+ fun:*
+ fun:_dl_allocate_tls_init
+ fun:dl_main
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Cond
+ fun:__tls*
+ obj:*
+ obj:*
+ fun:_vgnU_freeres
+}
+
+{
+ <glibc-2.10 overreads/conditionals>
+ Memcheck:Param
+ arch_prctl(arg2)
+ fun:init_tls
+}
+# GLib caching tmp/home directories (glibc-2.10 variants)
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Cond
+ fun:*
+ fun:dl_open_worker
+ fun:*
+ fun:*
+ fun:*
+ fun:_dl_catch_error
+ fun:dlerror_run
+ fun:*
+ fun:__nss_lookup_function
+ fun:__nss_lookup
+ fun:getpwnam*
+}
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Value8
+ fun:*
+ fun:dl_open_worker
+ fun:*
+ fun:*
+ fun:*
+ fun:_dl_catch_error
+ fun:dlerror_run
+ fun:*
+ fun:__nss_lookup_function
+ fun:__nss_lookup
+ fun:getpwnam*
+}
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Cond
+ fun:dl_open_worker
+ fun:*
+ fun:*
+ fun:do_dlopen
+ fun:*
+ fun:dlerror_run
+ fun:*
+ fun:__nss_lookup_function
+ fun:__nss_lookup
+ fun:getpwnam*
+}
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Value8
+ fun:dl_open_worker
+ fun:*
+ fun:*
+ fun:do_dlopen
+ fun:*
+ fun:dlerror_run
+ fun:*
+ fun:__nss_lookup_function
+ fun:__nss_lookup
+ fun:getpwnam*
+}
+
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Value8
+ fun:_dl_add_to_slotinfo
+ fun:dl_main
+}
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Param
+ open(filename)
+ fun:open
+ fun:open_verify
+ fun:open_path
+ fun:_dl_map_object
+}
+
+
+
+# GModule issues with glibc-2.10
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Value8
+ fun:*
+ fun:*
+ fun:dlsym
+ fun:g_module_symbol
+}
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Value8
+ fun:g_module_*
+ fun:gst_plugin*
+}
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Value8
+ fun:*
+ fun:g_module_*
+ fun:gst_plugin*
+}
+
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Value8
+ fun:*
+ fun:*
+ fun:dlopen*
+ fun:g_module_open
+}
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Value8
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:dlsym
+ fun:g_module_symbol
+}
+
+{
+ <glibc-2.10 GLIB leaks>
+ Memcheck:Value8
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:*
+ fun:dlopen*
+ fun:g_module_open
+}
+
+# Leak in GSlice
+{
+ <insert a suppression name here>
+ Memcheck:Value8
+ fun:g_parse_debug_string
+ fun:slice_config_init
+ fun:g_slice_init_nomessage
+ fun:_g_slice_thread_init_nomessage
+ fun:g_thread_init_glib
+}
+
+# 2.10 pthread issues
+{
+ <insert a suppression name here>
+ Memcheck:Value8
+ fun:__pthread_initialize_minimal
+}
+
+# glibc 2.11 conditional
+{
+ <glibc-2.11 conditional>
+ Memcheck:Cond
+ fun:_dl_relocate_object
+ fun:dl_main
+ fun:_dl_sysdep_start
+ fun:_dl_start
+ obj:/lib64/ld-2.11.so
+}
+
+# glibc 2.11 Leak
+
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ fun:_dl_*
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ fun:_dl_*
+ fun:_dl_*
+ fun:_dl_*
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:_dl_*
+ fun:openaux
+ fun:_dl_catch_error
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:_dl_map_object
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ fun:_dl_new_object
+ fun:_dl_map_object_from_fd
+ fun:_dl_map_object
+ fun:openaux
+ fun:_dl_catch_error
+ fun:_dl_map_object_deps
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:_dl_*
+ fun:_dl_*
+ fun:_dl_*
+ fun:dl_open_worker
+ fun:_dl_catch_error
+ fun:_dl_open
+ fun:dlopen_doit
+ fun:_dl_catch_error
+ fun:_dlerror_run
+ fun:dlopen@@GLIBC_2.2.5
+}
+
+# glib type leaks
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:g_type_register_static
+}
+
+# new registry system
+# all of this will only be created once when loading registry.
+
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:_priv_gst_registry_chunks_load_plugin
+}
+
+# system-wide tags
+# these tags are registered once
+
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ fun:*
+ fun:*
+ fun:gst_tag_register
+ fun:_gst_tag_initialize
+}
+
+# system-wide type classes that we keep referenced
+
+{
+ <g_type_class_ref leaks>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:g_type_class_ref
+}
+
+# leaking cached queries which are only initialized once
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:_gst_query_initialize
+ fun:init_post
+}
+
+# macosx (leopard) library loader leak
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:_Znwm
+ fun:_ZNSs4_Rep9_S_createEmmRKSaIcE
+ fun:_ZNSs12_S_constructIPKcEEPcT_S3_RKSaIcESt20forward_iterator_tag
+ fun:_ZNSsC2EPKcRKSaIcE
+ fun:_Z41__static_initialization_and_destruction_0ii
+ fun:_ZN16ImageLoaderMachO18doModInitFunctionsERKN11ImageLoader11LinkContextE
+}
+
+# GObject type registration
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:_g_atomic_array_copy
+}
+
+{
+ <getdelim one-time inits called from libselinux>
+ Memcheck:Leak
+ fun:*alloc
+ fun:getdelim
+ obj:*libselinux*
+}
+
+{
+ <weird one when re-reading registry>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ obj:*/sed
+}
+
+{
+ <weird one when re-reading registry>
+ Memcheck:Addr8
+ ...
+ obj:*/sed
+}
+
+# GLib 2.23 interface vtable
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:g_type_add_interface_static
+}
+
+{
+ <leak in dash on debian sid>
+ Memcheck:Leak
+ fun:*alloc
+ obj:*/dash
+}
+
+# libtool/gentoo fake leak
+# it actually runs bash and valgrind complains
+{
+ <insert_a_suppression_name_here>
+ Memcheck:Leak
+ fun:*alloc
+ obj:/bin/bash
+}
+
+{
+ <ignore possbly-lost leaks in the plugin scanner which doesn't clean up properly>
+ Memcheck:Leak
+ fun:*alloc
+ ...
+ fun:_gst_plugin_loader_client_run
+ fun:main
+}
+
+{
+ <warning with libc 2.13-2 as in Debian/unstable on amd64>
+ Memcheck:Cond
+ fun:*strcasecmp*
+ ...
+ fun:__dcigettext
+}
+
+{
+ <warning with libc 2.13-2 as in Debian/unstable on amd64>
+ Memcheck:Value8
+ fun:*strcasecmp*
+ ...
+ fun:__dcigettext
+}
+
+{
+ <GstSystemClock is a singleton and does not leak>
+ Memcheck:Leak
+ fun:malloc
+ ...
+ fun:gst_poll_new
+ fun:gst_poll_new_timer
+ fun:gst_system_clock_init
+}
+
+{
+ <glib types are singletons>
+ Memcheck:Leak
+ fun:calloc
+ ...
+ fun:gobject_init_ctor
+}
+
+{
+ <quark table is leaked on purpose if it grows too big>
+ Memcheck:Leak
+ fun:malloc
+ ...
+ fun:g_quark_from*_string
+}
--- /dev/null
+#!/usr/bin/env perl
+# -*- cperl -*-
+#
+# gtk-doc - GTK DocBook documentation generator.
+# Copyright (C) 1998 Damon Chaplin
+#
+# This program is free software; you can redistribute it and/or modify
+# it under the terms of the GNU General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA.
+#
+
+#
+# This gets information about object hierarchies and signals
+# by compiling a small C program. CFLAGS and LDFLAGS must be
+# set appropriately before running this script.
+#
+
+use Getopt::Long;
+
+my $GTK_DOC_PREFIX=`pkg-config --variable prefix gtk-doc`;
+if ($GTK_DOC_PREFIX) {
+ chomp $GTK_DOC_PREFIX;
+ #print "Adding $GTK_DOC_PREFIX/share/gtk-doc/data to \@INC\n";
+ unshift @INC, "$GTK_DOC_PREFIX/share/gtk-doc/data";
+} else {
+ unshift @INC, '/usr/share/gtk-doc/data';
+}
+require "gtkdoc-common.pl";
+
+# Options
+
+# name of documentation module
+my $MODULE;
+my $OUTPUT_DIR;
+my $INSPECT_DIR;
+my $VERBOSE;
+my $PRINT_VERSION;
+my $PRINT_HELP;
+my $TYPE_INIT_FUNC="g_type_init ()";
+
+# --nogtkinit is deprecated, as it is the default now anyway.
+%optctl = (module => \$MODULE,
+ source => \$SOURCE,
+ types => \$TYPES_FILE,
+ nogtkinit => \$NO_GTK_INIT,
+ 'type-init-func' => \$TYPE_INIT_FUNC,
+ 'output-dir' => \$OUTPUT_DIR,
+ 'inspect-dir' => \$INSPECT_DIR,
+ 'verbose' => \$VERBOSE,
+ 'version' => \$PRINT_VERSION,
+ 'help' => \$PRINT_HELP);
+
+GetOptions(\%optctl, "module=s", "source=s", "types:s", "output-dir:s", "inspect-dir:s", "nogtkinit", "type-init-func:s", "verbose", "version", "help");
+
+if ($NO_GTK_INIT) {
+ # Do nothing. This just avoids a warning.
+ # the option is not used anymore
+}
+
+if ($PRINT_VERSION) {
+ print "1.5\n";
+ exit 0;
+}
+
+if (!$MODULE) {
+ $PRINT_HELP = 1;
+}
+
+if ($PRINT_HELP) {
+ print <<EOF;
+gstdoc-scangobj version 1.5 - introspect gstreamer-plugins
+
+--module=MODULE_NAME Name of the doc module being parsed
+--source=SOURCE_NAME Name of the source module for plugins
+--types=FILE The name of the file to store the types in
+--type-init-func=FUNC The init function to call instead of g_type_init()
+--output-dir=DIRNAME The directory where the results are stored
+--inspect-dir=DIRNAME The directory where the plugin inspect data is stored
+--verbose Print extra output while processing
+--version Print the version of this program
+--help Print this help
+EOF
+ exit 0;
+}
+
+$OUTPUT_DIR = $OUTPUT_DIR ? $OUTPUT_DIR : ".";
+
+$TYPES_FILE = $TYPES_FILE ? $TYPES_FILE : "$OUTPUT_DIR/$MODULE.types";
+
+open (TYPES, $TYPES_FILE) || die "Cannot open $TYPES_FILE: $!\n";
+open (OUTPUT, ">$MODULE-scan.c") || die "Cannot open $MODULE-scan.c: $!\n";
+
+my $old_signals_filename = "$OUTPUT_DIR/$MODULE.signals";
+my $new_signals_filename = "$OUTPUT_DIR/$MODULE.signals.new";
+my $old_hierarchy_filename = "$OUTPUT_DIR/$MODULE.hierarchy";
+my $new_hierarchy_filename = "$OUTPUT_DIR/$MODULE.hierarchy.new";
+my $old_interfaces_filename = "$OUTPUT_DIR/$MODULE.interfaces";
+my $new_interfaces_filename = "$OUTPUT_DIR/$MODULE.interfaces.new";
+my $old_prerequisites_filename = "$OUTPUT_DIR/$MODULE.prerequisites";
+my $new_prerequisites_filename = "$OUTPUT_DIR/$MODULE.prerequisites.new";
+my $old_args_filename = "$OUTPUT_DIR/$MODULE.args";
+my $new_args_filename = "$OUTPUT_DIR/$MODULE.args.new";
+
+my $debug_log="g_message";
+if (!defined($VERBOSE) or $VERBOSE eq "0") {
+ $debug_log="//$debug_log";
+}
+
+# write a C program to scan the types
+
+$includes = "";
+@types = ();
+@impl_types = ();
+
+for (<TYPES>) {
+ if (/^#include/) {
+ $includes .= $_;
+ } elsif (/^%/) {
+ next;
+ } elsif (/^\s*$/) {
+ next;
+ } elsif (/^type:(.*)$/) {
+ $t = $1;
+ chomp $t;
+ push @impl_types, $t;
+ } else {
+ chomp;
+ push @types, $_;
+ }
+}
+
+$ntypes = @types + @impl_types + 1;
+
+print OUTPUT <<EOT;
+
+/* file generated by common/gstdoc-scangobj */
+
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <errno.h>
+
+#include <gst/gst.h>
+EOT
+
+if ($includes) {
+ print OUTPUT $includes;
+} else {
+ for (@types) {
+ print OUTPUT "extern GType $_ (void);\n";
+ }
+}
+
+print OUTPUT <<EOT;
+
+#ifdef GTK_IS_WIDGET_CLASS
+#include <gtk/gtkversion.h>
+#endif
+
+static GType *object_types = NULL;
+
+static GString *xmlstr = NULL;
+
+static const gchar*
+xmlprint (gint indent, const gchar *tag, const gchar *data)
+{
+ const gchar indent_str[] = " ";
+
+ /* reset */
+ g_string_truncate (xmlstr, 0);
+ g_string_append_len (xmlstr, indent_str, MIN (indent, strlen (indent_str)));
+ g_string_append_printf (xmlstr, "<%s>", tag);
+
+ if (data) {
+ gchar *s;
+
+ s = g_markup_escape_text (data, -1);
+ g_string_append (xmlstr, s);
+ g_free (s);
+ }
+
+ g_string_append_printf (xmlstr, "</%s>\\n", tag);
+ return xmlstr->str;
+}
+
+static gint
+gst_feature_sort_compare (gconstpointer a, gconstpointer b)
+{
+ const gchar *name_a = gst_plugin_feature_get_name ((GstPluginFeature *) a);
+ const gchar *name_b = gst_plugin_feature_get_name ((GstPluginFeature *) b);
+ return strcmp (name_a, name_b);
+}
+
+static gint
+static_pad_template_compare (gconstpointer a, gconstpointer b)
+{
+ GstStaticPadTemplate *spt_a = (GstStaticPadTemplate *) a;
+ GstStaticPadTemplate *spt_b = (GstStaticPadTemplate *) b;
+
+ /* we want SINK before SRC (enum is UNKNOWN, SRC, SINK) */
+ if (spt_a->direction != spt_b->direction)
+ return spt_b->direction - spt_a->direction;
+
+ /* we want ALWAYS first, SOMETIMES second, REQUEST last
+ * (enum is ALWAYS, SOMETIMES, REQUEST) */
+ if (spt_a->presence != spt_b->presence)
+ return spt_a->presence - spt_b->presence;
+
+ return strcmp (spt_a->name_template, spt_b->name_template);
+}
+
+static GType *
+get_object_types (void)
+{
+ gpointer g_object_class;
+ GList *plugins = NULL;
+ GList *factories = NULL;
+ GList *l;
+ GstElementFactory *factory = NULL;
+ GType type;
+ gint i = 0;
+ gboolean reinspect;
+
+ /* get a list of features from plugins in our source module */
+ plugins = gst_registry_get_plugin_list (gst_registry_get ());
+
+ xmlstr = g_string_new ("");
+
+ reinspect = !g_file_test ("scanobj-build.stamp", G_FILE_TEST_EXISTS);
+
+ while (plugins) {
+ GList *features;
+ GstPlugin *plugin;
+ const gchar *source;
+ FILE *inspect = NULL;
+ gchar *inspect_name;
+
+ plugin = (GstPlugin *) (plugins->data);
+ plugins = g_list_next (plugins);
+ source = gst_plugin_get_source (plugin);
+ if (!source || strcmp (source, "$SOURCE") != 0) {
+ continue;
+ }
+
+ /* skip static coreelements plugin with pipeline and bin element factory */
+ if (gst_plugin_get_filename (plugin) == NULL)
+ continue;
+
+ $debug_log ("plugin: %s source: %s", gst_plugin_get_name (plugin), source);
+
+ if (reinspect) {
+ gchar *basename;
+
+ inspect_name = g_strdup_printf ("$INSPECT_DIR" G_DIR_SEPARATOR_S "plugin-%s.xml",
+ gst_plugin_get_name (plugin));
+ inspect = fopen (inspect_name, "w");
+ if (inspect == NULL) {
+ g_error ("Could not open %s for writing: %s\\n", inspect_name,
+ g_strerror (errno));
+ }
+ g_free (inspect_name);
+
+ basename = g_path_get_basename (gst_plugin_get_filename (plugin));
+
+ /* output plugin data */
+ fputs ("<plugin>\\n",inspect);
+ fputs (xmlprint(2, "name", gst_plugin_get_name (plugin)),inspect);
+ fputs (xmlprint(2, "description", gst_plugin_get_description (plugin)),inspect);
+ fputs (xmlprint(2, "filename", gst_plugin_get_filename (plugin)),inspect);
+ fputs (xmlprint(2, "basename", basename),inspect);
+ fputs (xmlprint(2, "version", gst_plugin_get_version (plugin)),inspect);
+ fputs (xmlprint(2, "license", gst_plugin_get_license (plugin)),inspect);
+ fputs (xmlprint(2, "source", gst_plugin_get_source (plugin)),inspect);
+ fputs (xmlprint(2, "package", gst_plugin_get_package (plugin)),inspect);
+ fputs (xmlprint(2, "origin", gst_plugin_get_origin (plugin)),inspect);
+ fputs (" <elements>\\n", inspect);
+
+ g_free (basename);
+ }
+
+ features =
+ gst_registry_get_feature_list_by_plugin (gst_registry_get (),
+ gst_plugin_get_name (plugin));
+
+ /* sort factories by feature->name */
+ features = g_list_sort (features, gst_feature_sort_compare);
+
+ while (features) {
+ GstPluginFeature *feature;
+ feature = GST_PLUGIN_FEATURE (features->data);
+ feature = gst_plugin_feature_load (feature);
+ if (!feature) {
+ g_warning ("Could not load plugin feature %s",
+ gst_plugin_feature_get_name (feature));
+ }
+
+ if (GST_IS_ELEMENT_FACTORY (feature)) {
+ const gchar *pad_dir[] = { "unknown","source","sink" };
+ const gchar *pad_pres[] = { "always","sometimes","request" };
+ GList *pads, *pad;
+
+ $debug_log (" feature: %s", gst_plugin_feature_get_name (feature));
+
+ factory = GST_ELEMENT_FACTORY (feature);
+ factories = g_list_prepend (factories, factory);
+
+ if (reinspect) {
+ /* output element data */
+ fputs (" <element>\\n", inspect);
+ fputs (xmlprint(6, "name", gst_plugin_feature_get_name (feature)),inspect);
+ fputs (xmlprint(6, "longname", gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_LONGNAME)),inspect);
+ fputs (xmlprint(6, "class", gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_KLASS)),inspect);
+ fputs (xmlprint(6, "description", gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_DESCRIPTION)),inspect);
+ fputs (xmlprint(6, "author", gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_AUTHOR)),inspect);
+ fputs (" <pads>\\n", inspect);
+
+ /* output pad-template data */
+ pads = g_list_copy ((GList *) gst_element_factory_get_static_pad_templates (factory));
+ pads = g_list_sort (pads, static_pad_template_compare);
+ for (pad = pads; pad != NULL; pad = pad->next) {
+ GstStaticPadTemplate *pt = pad->data;
+
+ fputs (" <caps>\\n", inspect);
+ fputs (xmlprint(10, "name", pt->name_template),inspect);
+ fputs (xmlprint(10, "direction", pad_dir[pt->direction]),inspect);
+ fputs (xmlprint(10, "presence", pad_pres[pt->presence]),inspect);
+ fputs (xmlprint(10, "details", pt->static_caps.string),inspect);
+ fputs (" </caps>\\n", inspect);
+ }
+ g_list_free (pads);
+ fputs (" </pads>\\n </element>\\n", inspect);
+ }
+ }
+ features = g_list_next (features);
+ }
+
+ if (reinspect) {
+ fputs (" </elements>\\n</plugin>", inspect);
+ fclose (inspect);
+ }
+ }
+
+ g_string_free (xmlstr, TRUE);
+
+ $debug_log ("number of element factories: %d", g_list_length (factories));
+
+ /* allocate the object_types array to hold them */
+ object_types = g_new0 (GType, g_list_length (factories)+$ntypes+1);
+
+ l = factories;
+ i = 0;
+
+ /* fill it */
+ while (l) {
+ factory = GST_ELEMENT_FACTORY (l->data);
+ type = gst_element_factory_get_element_type (factory);
+ if (type != 0) {
+ $debug_log ("adding type for factory %s", gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_LONGNAME));
+ object_types[i++] = type;
+ } else {
+ g_message ("type info for factory %s not found",
+ gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_LONGNAME));
+ }
+ l = g_list_next (l);
+ }
+
+EOT
+
+# get_type functions:
+for (@types) {
+print OUTPUT <<EOT;
+ type = $_ ();
+ if (type == 0) {
+ g_message ("$_ () didn't return a valid type");
+ }
+ else {
+ object_types[i++] = type;
+ }
+EOT
+}
+
+# Implicit types retrieved from GLib:
+for (@impl_types) {
+print OUTPUT <<EOT;
+ type = g_type_from_name ("$_");
+ if (type == 0) {
+ g_message ("Implicit type $_ not found");
+ }
+ else {
+ object_types[i++] = type;
+ }
+EOT
+}
+
+print OUTPUT <<EOT;
+
+ object_types[i] = 0;
+
+ /* reference the GObjectClass to initialize the param spec pool
+ * potentially needed by interfaces. See http://bugs.gnome.org/571820 */
+ g_object_class = g_type_class_ref (G_TYPE_OBJECT);
+
+ /* Need to make sure all the types are loaded in and initialize
+ * their signals and properties.
+ */
+ for (i=0; object_types[i]; i++)
+ {
+ if (G_TYPE_IS_CLASSED (object_types[i]))
+ g_type_class_ref (object_types[i]);
+ if (G_TYPE_IS_INTERFACE (object_types[i]))
+ g_type_default_interface_ref (object_types[i]);
+ }
+
+ g_type_class_unref (g_object_class);
+
+ return object_types;
+}
+
+/*
+ * This uses GObject type functions to output signal prototypes and the object
+ * hierarchy.
+ */
+
+/* The output files */
+const gchar *signals_filename = "$new_signals_filename";
+const gchar *hierarchy_filename = "$new_hierarchy_filename";
+const gchar *interfaces_filename = "$new_interfaces_filename";
+const gchar *prerequisites_filename = "$new_prerequisites_filename";
+const gchar *args_filename = "$new_args_filename";
+
+
+static void output_signals (void);
+static void output_object_signals (FILE *fp,
+ GType object_type);
+static void output_object_signal (FILE *fp,
+ const gchar *object_class_name,
+ guint signal_id);
+static const gchar * get_type_name (GType type,
+ gboolean * is_pointer);
+static void output_object_hierarchy (void);
+static void output_hierarchy (FILE *fp,
+ GType type,
+ guint level);
+
+static void output_object_interfaces (void);
+static void output_interfaces (FILE *fp,
+ GType type);
+
+static void output_interface_prerequisites (void);
+static void output_prerequisites (FILE *fp,
+ GType type);
+
+static void output_args (void);
+static void output_object_args (FILE *fp, GType object_type);
+
+int
+main (G_GNUC_UNUSED int argc, G_GNUC_UNUSED char *argv[])
+{
+ $TYPE_INIT_FUNC;
+
+ get_object_types ();
+
+ output_signals ();
+ output_object_hierarchy ();
+ output_object_interfaces ();
+ output_interface_prerequisites ();
+ output_args ();
+
+ return 0;
+}
+
+
+static void
+output_signals (void)
+{
+ FILE *fp;
+ gint i;
+
+ fp = fopen (signals_filename, "w");
+ if (fp == NULL)
+ {
+ g_warning ("Couldn't open output file: %s : %s", signals_filename, g_strerror(errno));
+ return;
+ }
+
+ for (i = 0; object_types[i]; i++)
+ output_object_signals (fp, object_types[i]);
+
+ fclose (fp);
+}
+
+static gint
+compare_signals (const void *a, const void *b)
+{
+ const guint *signal_a = a;
+ const guint *signal_b = b;
+
+ return strcmp (g_signal_name (*signal_a), g_signal_name (*signal_b));
+}
+
+/* This outputs all the signals of one object. */
+static void
+output_object_signals (FILE *fp, GType object_type)
+{
+ const gchar *object_class_name;
+ guint *signals, n_signals;
+ guint sig;
+
+ if (G_TYPE_IS_INSTANTIATABLE (object_type) ||
+ G_TYPE_IS_INTERFACE (object_type))
+ {
+
+ object_class_name = g_type_name (object_type);
+
+ signals = g_signal_list_ids (object_type, &n_signals);
+ qsort (signals, n_signals, sizeof (guint), compare_signals);
+
+ for (sig = 0; sig < n_signals; sig++)
+ {
+ output_object_signal (fp, object_class_name, signals[sig]);
+ }
+ g_free (signals);
+ }
+}
+
+
+/* This outputs one signal. */
+static void
+output_object_signal (FILE *fp,
+ const gchar *object_name,
+ guint signal_id)
+{
+ GSignalQuery query_info;
+ const gchar *type_name, *ret_type, *object_arg, *arg_name;
+ gchar *pos, *object_arg_lower;
+ gboolean is_pointer;
+ gchar buffer[1024];
+ guint i, param;
+ gint param_num, widget_num, event_num, callback_num;
+ gint *arg_num;
+ gchar signal_name[128];
+ gchar flags[16];
+
+ $debug_log ("Object: %s Signal: %u", object_name, signal_id);
+
+ param_num = 1;
+ widget_num = event_num = callback_num = 0;
+
+ g_signal_query (signal_id, &query_info);
+
+ /* Output the signal object type and the argument name. We assume the
+ type is a pointer - I think that is OK. We remove "Gtk" or "Gnome" and
+ convert to lower case for the argument name. */
+ pos = buffer;
+ sprintf (pos, "%s ", object_name);
+ pos += strlen (pos);
+
+ /* Try to come up with a sensible variable name for the first arg
+ * It chops off 2 know prefixes :/ and makes the name lowercase
+ * It should replace lowercase -> uppercase with '_'
+ * GFileMonitor -> file_monitor
+ * GIOExtensionPoint -> extension_point
+ * GtkTreeView -> tree_view
+ * if 2nd char is upper case too
+ * search for first lower case and go back one char
+ * else
+ * search for next upper case
+ */
+ if (!strncmp (object_name, "Gtk", 3))
+ object_arg = object_name + 3;
+ else if (!strncmp (object_name, "Gnome", 5))
+ object_arg = object_name + 5;
+ else
+ object_arg = object_name;
+
+ object_arg_lower = g_ascii_strdown (object_arg, -1);
+ sprintf (pos, "*%s\\n", object_arg_lower);
+ pos += strlen (pos);
+ if (!strncmp (object_arg_lower, "widget", 6))
+ widget_num = 2;
+ g_free(object_arg_lower);
+
+ /* Convert signal name to use underscores rather than dashes '-'. */
+ strncpy (signal_name, query_info.signal_name, 127);
+ signal_name[127] = '\\0';
+ for (i = 0; signal_name[i]; i++)
+ {
+ if (signal_name[i] == '-')
+ signal_name[i] = '_';
+ }
+
+ /* Output the signal parameters. */
+ for (param = 0; param < query_info.n_params; param++)
+ {
+ type_name = get_type_name (query_info.param_types[param] & ~G_SIGNAL_TYPE_STATIC_SCOPE, &is_pointer);
+
+ /* Most arguments to the callback are called "arg1", "arg2", etc.
+ GtkWidgets are called "widget", "widget2", ...
+ GtkCallbacks are called "callback", "callback2", ... */
+ if (!strcmp (type_name, "GtkWidget"))
+ {
+ arg_name = "widget";
+ arg_num = &widget_num;
+ }
+ else if (!strcmp (type_name, "GtkCallback")
+ || !strcmp (type_name, "GtkCCallback"))
+ {
+ arg_name = "callback";
+ arg_num = &callback_num;
+ }
+ else
+ {
+ arg_name = "arg";
+ arg_num = ¶m_num;
+ }
+ sprintf (pos, "%s ", type_name);
+ pos += strlen (pos);
+
+ if (!arg_num || *arg_num == 0)
+ sprintf (pos, "%s%s\\n", is_pointer ? "*" : " ", arg_name);
+ else
+ sprintf (pos, "%s%s%i\\n", is_pointer ? "*" : " ", arg_name,
+ *arg_num);
+ pos += strlen (pos);
+
+ if (arg_num)
+ {
+ if (*arg_num == 0)
+ *arg_num = 2;
+ else
+ *arg_num += 1;
+ }
+ }
+
+ pos = flags;
+ /* We use one-character flags for simplicity. */
+ if (query_info.signal_flags & G_SIGNAL_RUN_FIRST)
+ *pos++ = 'f';
+ if (query_info.signal_flags & G_SIGNAL_RUN_LAST)
+ *pos++ = 'l';
+ if (query_info.signal_flags & G_SIGNAL_RUN_CLEANUP)
+ *pos++ = 'c';
+ if (query_info.signal_flags & G_SIGNAL_NO_RECURSE)
+ *pos++ = 'r';
+ if (query_info.signal_flags & G_SIGNAL_DETAILED)
+ *pos++ = 'd';
+ if (query_info.signal_flags & G_SIGNAL_ACTION)
+ *pos++ = 'a';
+ if (query_info.signal_flags & G_SIGNAL_NO_HOOKS)
+ *pos++ = 'h';
+ *pos = 0;
+
+ /* Output the return type and function name. */
+ ret_type = get_type_name (query_info.return_type & ~G_SIGNAL_TYPE_STATIC_SCOPE, &is_pointer);
+
+ fprintf (fp,
+ "<SIGNAL>\\n<NAME>%s::%s</NAME>\\n<RETURNS>%s%s</RETURNS>\\n<FLAGS>%s</FLAGS>\\n%s</SIGNAL>\\n\\n",
+ object_name, query_info.signal_name, ret_type, is_pointer ? "*" : "", flags, buffer);
+}
+
+
+/* Returns the type name to use for a signal argument or return value, given
+ the GtkType from the signal info. It also sets is_pointer to TRUE if the
+ argument needs a '*' since it is a pointer. */
+static const gchar *
+get_type_name (GType type, gboolean * is_pointer)
+{
+ const gchar *type_name;
+
+ *is_pointer = FALSE;
+ type_name = g_type_name (type);
+
+ switch (type) {
+ case G_TYPE_NONE:
+ case G_TYPE_CHAR:
+ case G_TYPE_UCHAR:
+ case G_TYPE_BOOLEAN:
+ case G_TYPE_INT:
+ case G_TYPE_UINT:
+ case G_TYPE_LONG:
+ case G_TYPE_ULONG:
+ case G_TYPE_FLOAT:
+ case G_TYPE_DOUBLE:
+ case G_TYPE_POINTER:
+ /* These all have normal C type names so they are OK. */
+ return type_name;
+
+ case G_TYPE_STRING:
+ /* A GtkString is really a gchar*. */
+ *is_pointer = TRUE;
+ return "gchar";
+
+ case G_TYPE_ENUM:
+ case G_TYPE_FLAGS:
+ /* We use a gint for both of these. Hopefully a subtype with a decent
+ name will be registered and used instead, as GTK+ does itself. */
+ return "gint";
+
+ case G_TYPE_BOXED:
+ /* The boxed type shouldn't be used itself, only subtypes. Though we
+ return 'gpointer' just in case. */
+ return "gpointer";
+
+ case G_TYPE_PARAM:
+ /* A GParam is really a GParamSpec*. */
+ *is_pointer = TRUE;
+ return "GParamSpec";
+
+#if GLIB_CHECK_VERSION (2, 25, 9)
+ case G_TYPE_VARIANT:
+ *is_pointer = TRUE;
+ return "GVariant";
+#endif
+
+default:
+ break;
+ }
+
+ /* For all GObject subclasses we can use the class name with a "*",
+ e.g. 'GtkWidget *'. */
+ if (g_type_is_a (type, G_TYPE_OBJECT))
+ *is_pointer = TRUE;
+
+ /* Also catch non GObject root types */
+ if (G_TYPE_IS_CLASSED (type))
+ *is_pointer = TRUE;
+
+ /* All boxed subtypes will be pointers as well. */
+ /* Exception: GStrv */
+ if (g_type_is_a (type, G_TYPE_BOXED) &&
+ !g_type_is_a (type, G_TYPE_STRV))
+ *is_pointer = TRUE;
+
+ /* All pointer subtypes will be pointers as well. */
+ if (g_type_is_a (type, G_TYPE_POINTER))
+ *is_pointer = TRUE;
+
+ /* But enums are not */
+ if (g_type_is_a (type, G_TYPE_ENUM) ||
+ g_type_is_a (type, G_TYPE_FLAGS))
+ *is_pointer = FALSE;
+
+ return type_name;
+}
+
+
+/* This outputs the hierarchy of all objects which have been initialized,
+ i.e. by calling their XXX_get_type() initialization function. */
+static void
+output_object_hierarchy (void)
+{
+ FILE *fp;
+ gint i,j;
+ GType root, type;
+ GType root_types[$ntypes] = { G_TYPE_INVALID, };
+
+ fp = fopen (hierarchy_filename, "w");
+ if (fp == NULL)
+ {
+ g_warning ("Couldn't open output file: %s : %s", hierarchy_filename, g_strerror(errno));
+ return;
+ }
+ output_hierarchy (fp, G_TYPE_OBJECT, 0);
+ output_hierarchy (fp, G_TYPE_INTERFACE, 0);
+
+ for (i=0; object_types[i]; i++) {
+ root = object_types[i];
+ while ((type = g_type_parent (root))) {
+ root = type;
+ }
+ if ((root != G_TYPE_OBJECT) && (root != G_TYPE_INTERFACE)) {
+ for (j=0; root_types[j]; j++) {
+ if (root == root_types[j]) {
+ root = G_TYPE_INVALID; break;
+ }
+ }
+ if(root) {
+ root_types[j] = root;
+ output_hierarchy (fp, root, 0);
+ }
+ }
+ }
+
+ fclose (fp);
+}
+
+static int
+compare_types (const void *a, const void *b)
+{
+ const char *na = g_type_name (*((GType *)a));
+ const char *nb = g_type_name (*((GType *)b));
+
+ return g_strcmp0 (na, nb);
+}
+
+
+/* This is called recursively to output the hierarchy of a object. */
+static void
+output_hierarchy (FILE *fp,
+ GType type,
+ guint level)
+{
+ guint i;
+ GType *children;
+ guint n_children;
+
+ if (!type)
+ return;
+
+ for (i = 0; i < level; i++)
+ fprintf (fp, " ");
+ fprintf (fp, "%s\\n", g_type_name (type));
+
+ children = g_type_children (type, &n_children);
+ qsort (children, n_children, sizeof (GType), compare_types);
+
+
+ for (i=0; i < n_children; i++)
+ output_hierarchy (fp, children[i], level + 1);
+
+ g_free (children);
+}
+
+static void output_object_interfaces (void)
+{
+ guint i;
+ FILE *fp;
+
+ fp = fopen (interfaces_filename, "w");
+ if (fp == NULL)
+ {
+ g_warning ("Couldn't open output file: %s : %s", interfaces_filename, g_strerror(errno));
+ return;
+ }
+ output_interfaces (fp, G_TYPE_OBJECT);
+
+ for (i = 0; object_types[i]; i++)
+ {
+ if (!g_type_parent (object_types[i]) &&
+ (object_types[i] != G_TYPE_OBJECT) &&
+ G_TYPE_IS_INSTANTIATABLE (object_types[i]))
+ {
+ output_interfaces (fp, object_types[i]);
+ }
+ }
+ fclose (fp);
+}
+
+static void
+output_interfaces (FILE *fp,
+ GType type)
+{
+ guint i;
+ GType *children, *interfaces;
+ guint n_children, n_interfaces;
+
+ if (!type)
+ return;
+
+ interfaces = g_type_interfaces (type, &n_interfaces);
+
+ if (n_interfaces > 0)
+ {
+ fprintf (fp, "%s", g_type_name (type));
+ for (i=0; i < n_interfaces; i++)
+ fprintf (fp, " %s", g_type_name (interfaces[i]));
+ fprintf (fp, "\\n");
+ }
+ g_free (interfaces);
+
+ children = g_type_children (type, &n_children);
+
+ for (i=0; i < n_children; i++)
+ output_interfaces (fp, children[i]);
+
+ g_free (children);
+}
+
+static void output_interface_prerequisites (void)
+{
+ FILE *fp;
+
+ fp = fopen (prerequisites_filename, "w");
+ if (fp == NULL)
+ {
+ g_warning ("Couldn't open output file: %s : %s", prerequisites_filename, g_strerror(errno));
+ return;
+ }
+ output_prerequisites (fp, G_TYPE_INTERFACE);
+ fclose (fp);
+}
+
+static void
+output_prerequisites (FILE *fp,
+ GType type)
+{
+#if GLIB_CHECK_VERSION(2,1,0)
+ guint i;
+ GType *children, *prerequisites;
+ guint n_children, n_prerequisites;
+
+ if (!type)
+ return;
+
+ prerequisites = g_type_interface_prerequisites (type, &n_prerequisites);
+
+ if (n_prerequisites > 0)
+ {
+ fprintf (fp, "%s", g_type_name (type));
+ for (i=0; i < n_prerequisites; i++)
+ fprintf (fp, " %s", g_type_name (prerequisites[i]));
+ fprintf (fp, "\\n");
+ }
+ g_free (prerequisites);
+
+ children = g_type_children (type, &n_children);
+
+ for (i=0; i < n_children; i++)
+ output_prerequisites (fp, children[i]);
+
+ g_free (children);
+#endif
+}
+
+static void
+output_args (void)
+{
+ FILE *fp;
+ gint i;
+
+ fp = fopen (args_filename, "w");
+ if (fp == NULL)
+ {
+ g_warning ("Couldn't open output file: %s : %s", args_filename, g_strerror(errno));
+ return;
+ }
+
+ for (i = 0; object_types[i]; i++) {
+ output_object_args (fp, object_types[i]);
+ }
+
+ fclose (fp);
+}
+
+static gint
+compare_param_specs (const void *a, const void *b)
+{
+ GParamSpec *spec_a = *(GParamSpec **)a;
+ GParamSpec *spec_b = *(GParamSpec **)b;
+
+ return strcmp (g_param_spec_get_name (spec_a), g_param_spec_get_name (spec_b));
+}
+
+/* Its common to have unsigned properties restricted
+ * to the signed range. Therefore we make this look
+ * a bit nicer by spelling out the max constants.
+ */
+
+/* Don't use "==" with floats, it might trigger a gcc warning. */
+#define GTKDOC_COMPARE_FLOAT(x, y) (x <= y && x >= y)
+
+static gchar*
+describe_double_constant (gdouble value)
+{
+ gchar *desc;
+
+ if (GTKDOC_COMPARE_FLOAT (value, G_MAXDOUBLE))
+ desc = g_strdup ("G_MAXDOUBLE");
+ else if (GTKDOC_COMPARE_FLOAT (value, G_MINDOUBLE))
+ desc = g_strdup ("G_MINDOUBLE");
+ else if (GTKDOC_COMPARE_FLOAT (value, -G_MAXDOUBLE))
+ desc = g_strdup ("-G_MAXDOUBLE");
+ else if (GTKDOC_COMPARE_FLOAT (value, G_MAXFLOAT))
+ desc = g_strdup ("G_MAXFLOAT");
+ else if (GTKDOC_COMPARE_FLOAT (value, G_MINFLOAT))
+ desc = g_strdup ("G_MINFLOAT");
+ else if (GTKDOC_COMPARE_FLOAT (value, -G_MAXFLOAT))
+ desc = g_strdup ("-G_MAXFLOAT");
+ else{
+ /* make sure floats are output with a decimal dot irrespective of
+ * current locale. Use formatd since we want human-readable numbers
+ * and do not need the exact same bit representation when deserialising */
+ desc = g_malloc0 (G_ASCII_DTOSTR_BUF_SIZE);
+ g_ascii_formatd (desc, G_ASCII_DTOSTR_BUF_SIZE, "%g", value);
+ }
+
+ return desc;
+}
+
+static gchar*
+describe_signed_constant (gsize size, gint64 value)
+{
+ gchar *desc = NULL;
+
+ switch (size) {
+ case 8:
+ if (value == G_MAXINT64)
+ desc = g_strdup ("G_MAXINT64");
+ else if (value == G_MININT64)
+ desc = g_strdup ("G_MININT64");
+ /* fall through */
+ case 4:
+ if (sizeof (int) == 4) {
+ if (value == G_MAXINT)
+ desc = g_strdup ("G_MAXINT");
+ else if (value == G_MININT)
+ desc = g_strdup ("G_MININT");
+ else if (value == (gint64)G_MAXUINT)
+ desc = g_strdup ("G_MAXUINT");
+ }
+ if (value == G_MAXLONG)
+ desc = g_strdup ("G_MAXLONG");
+ else if (value == G_MINLONG)
+ desc = g_strdup ("G_MINLONG");
+ else if (value == (gint64)G_MAXULONG)
+ desc = g_strdup ("G_MAXULONG");
+ /* fall through */
+ case 2:
+ if (sizeof (int) == 2) {
+ if (value == G_MAXINT)
+ desc = g_strdup ("G_MAXINT");
+ else if (value == G_MININT)
+ desc = g_strdup ("G_MININT");
+ else if (value == (gint64)G_MAXUINT)
+ desc = g_strdup ("G_MAXUINT");
+ }
+ break;
+ default:
+ break;
+ }
+ if (!desc)
+ desc = g_strdup_printf ("%" G_GINT64_FORMAT, value);
+
+ return desc;
+}
+
+static gchar*
+describe_unsigned_constant (gsize size, guint64 value)
+{
+ gchar *desc = NULL;
+
+ switch (size) {
+ case 8:
+ if (value == G_MAXINT64)
+ desc = g_strdup ("G_MAXINT64");
+ else if (value == G_MAXUINT64)
+ desc = g_strdup ("G_MAXUINT64");
+ /* fall through */
+ case 4:
+ if (sizeof (int) == 4) {
+ if (value == (guint64)G_MAXINT)
+ desc = g_strdup ("G_MAXINT");
+ else if (value == G_MAXUINT)
+ desc = g_strdup ("G_MAXUINT");
+ }
+ if (value == (guint64)G_MAXLONG)
+ desc = g_strdup ("G_MAXLONG");
+ else if (value == G_MAXULONG)
+ desc = g_strdup ("G_MAXULONG");
+ /* fall through */
+ case 2:
+ if (sizeof (int) == 2) {
+ if (value == (guint64)G_MAXINT)
+ desc = g_strdup ("G_MAXINT");
+ else if (value == G_MAXUINT)
+ desc = g_strdup ("G_MAXUINT");
+ }
+ break;
+ default:
+ break;
+ }
+ if (!desc)
+ desc = g_strdup_printf ("%" G_GUINT64_FORMAT, value);
+
+ return desc;
+}
+
+static gchar*
+describe_type (GParamSpec *spec)
+{
+ gchar *desc;
+ gchar *lower;
+ gchar *upper;
+
+ if (G_IS_PARAM_SPEC_CHAR (spec))
+ {
+ GParamSpecChar *pspec = G_PARAM_SPEC_CHAR (spec);
+
+ lower = describe_signed_constant (sizeof(gchar), pspec->minimum);
+ upper = describe_signed_constant (sizeof(gchar), pspec->maximum);
+ if (pspec->minimum == G_MININT8 && pspec->maximum == G_MAXINT8)
+ desc = g_strdup ("");
+ else if (pspec->minimum == G_MININT8)
+ desc = g_strdup_printf ("<= %s", upper);
+ else if (pspec->maximum == G_MAXINT8)
+ desc = g_strdup_printf (">= %s", lower);
+ else
+ desc = g_strdup_printf ("[%s,%s]", lower, upper);
+ g_free (lower);
+ g_free (upper);
+ }
+ else if (G_IS_PARAM_SPEC_UCHAR (spec))
+ {
+ GParamSpecUChar *pspec = G_PARAM_SPEC_UCHAR (spec);
+
+ lower = describe_unsigned_constant (sizeof(guchar), pspec->minimum);
+ upper = describe_unsigned_constant (sizeof(guchar), pspec->maximum);
+ if (pspec->minimum == 0 && pspec->maximum == G_MAXUINT8)
+ desc = g_strdup ("");
+ else if (pspec->minimum == 0)
+ desc = g_strdup_printf ("<= %s", upper);
+ else if (pspec->maximum == G_MAXUINT8)
+ desc = g_strdup_printf (">= %s", lower);
+ else
+ desc = g_strdup_printf ("[%s,%s]", lower, upper);
+ g_free (lower);
+ g_free (upper);
+ }
+ else if (G_IS_PARAM_SPEC_INT (spec))
+ {
+ GParamSpecInt *pspec = G_PARAM_SPEC_INT (spec);
+
+ lower = describe_signed_constant (sizeof(gint), pspec->minimum);
+ upper = describe_signed_constant (sizeof(gint), pspec->maximum);
+ if (pspec->minimum == G_MININT && pspec->maximum == G_MAXINT)
+ desc = g_strdup ("");
+ else if (pspec->minimum == G_MININT)
+ desc = g_strdup_printf ("<= %s", upper);
+ else if (pspec->maximum == G_MAXINT)
+ desc = g_strdup_printf (">= %s", lower);
+ else
+ desc = g_strdup_printf ("[%s,%s]", lower, upper);
+ g_free (lower);
+ g_free (upper);
+ }
+ else if (G_IS_PARAM_SPEC_UINT (spec))
+ {
+ GParamSpecUInt *pspec = G_PARAM_SPEC_UINT (spec);
+
+ lower = describe_unsigned_constant (sizeof(guint), pspec->minimum);
+ upper = describe_unsigned_constant (sizeof(guint), pspec->maximum);
+ if (pspec->minimum == 0 && pspec->maximum == G_MAXUINT)
+ desc = g_strdup ("");
+ else if (pspec->minimum == 0)
+ desc = g_strdup_printf ("<= %s", upper);
+ else if (pspec->maximum == G_MAXUINT)
+ desc = g_strdup_printf (">= %s", lower);
+ else
+ desc = g_strdup_printf ("[%s,%s]", lower, upper);
+ g_free (lower);
+ g_free (upper);
+ }
+ else if (G_IS_PARAM_SPEC_LONG (spec))
+ {
+ GParamSpecLong *pspec = G_PARAM_SPEC_LONG (spec);
+
+ lower = describe_signed_constant (sizeof(glong), pspec->minimum);
+ upper = describe_signed_constant (sizeof(glong), pspec->maximum);
+ if (pspec->minimum == G_MINLONG && pspec->maximum == G_MAXLONG)
+ desc = g_strdup ("");
+ else if (pspec->minimum == G_MINLONG)
+ desc = g_strdup_printf ("<= %s", upper);
+ else if (pspec->maximum == G_MAXLONG)
+ desc = g_strdup_printf (">= %s", lower);
+ else
+ desc = g_strdup_printf ("[%s,%s]", lower, upper);
+ g_free (lower);
+ g_free (upper);
+ }
+ else if (G_IS_PARAM_SPEC_ULONG (spec))
+ {
+ GParamSpecULong *pspec = G_PARAM_SPEC_ULONG (spec);
+
+ lower = describe_unsigned_constant (sizeof(gulong), pspec->minimum);
+ upper = describe_unsigned_constant (sizeof(gulong), pspec->maximum);
+ if (pspec->minimum == 0 && pspec->maximum == G_MAXULONG)
+ desc = g_strdup ("");
+ else if (pspec->minimum == 0)
+ desc = g_strdup_printf ("<= %s", upper);
+ else if (pspec->maximum == G_MAXULONG)
+ desc = g_strdup_printf (">= %s", lower);
+ else
+ desc = g_strdup_printf ("[%s,%s]", lower, upper);
+ g_free (lower);
+ g_free (upper);
+ }
+ else if (G_IS_PARAM_SPEC_INT64 (spec))
+ {
+ GParamSpecInt64 *pspec = G_PARAM_SPEC_INT64 (spec);
+
+ lower = describe_signed_constant (sizeof(gint64), pspec->minimum);
+ upper = describe_signed_constant (sizeof(gint64), pspec->maximum);
+ if (pspec->minimum == G_MININT64 && pspec->maximum == G_MAXINT64)
+ desc = g_strdup ("");
+ else if (pspec->minimum == G_MININT64)
+ desc = g_strdup_printf ("<= %s", upper);
+ else if (pspec->maximum == G_MAXINT64)
+ desc = g_strdup_printf (">= %s", lower);
+ else
+ desc = g_strdup_printf ("[%s,%s]", lower, upper);
+ g_free (lower);
+ g_free (upper);
+ }
+ else if (G_IS_PARAM_SPEC_UINT64 (spec))
+ {
+ GParamSpecUInt64 *pspec = G_PARAM_SPEC_UINT64 (spec);
+
+ lower = describe_unsigned_constant (sizeof(guint64), pspec->minimum);
+ upper = describe_unsigned_constant (sizeof(guint64), pspec->maximum);
+ if (pspec->minimum == 0 && pspec->maximum == G_MAXUINT64)
+ desc = g_strdup ("");
+ else if (pspec->minimum == 0)
+ desc = g_strdup_printf ("<= %s", upper);
+ else if (pspec->maximum == G_MAXUINT64)
+ desc = g_strdup_printf (">= %s", lower);
+ else
+ desc = g_strdup_printf ("[%s,%s]", lower, upper);
+ g_free (lower);
+ g_free (upper);
+ }
+ else if (G_IS_PARAM_SPEC_FLOAT (spec))
+ {
+ GParamSpecFloat *pspec = G_PARAM_SPEC_FLOAT (spec);
+
+ lower = describe_double_constant (pspec->minimum);
+ upper = describe_double_constant (pspec->maximum);
+ if (GTKDOC_COMPARE_FLOAT (pspec->minimum, -G_MAXFLOAT))
+ {
+ if (GTKDOC_COMPARE_FLOAT (pspec->maximum, G_MAXFLOAT))
+ desc = g_strdup ("");
+ else
+ desc = g_strdup_printf ("<= %s", upper);
+ }
+ else if (GTKDOC_COMPARE_FLOAT (pspec->maximum, G_MAXFLOAT))
+ desc = g_strdup_printf (">= %s", lower);
+ else
+ desc = g_strdup_printf ("[%s,%s]", lower, upper);
+ g_free (lower);
+ g_free (upper);
+ }
+ else if (G_IS_PARAM_SPEC_DOUBLE (spec))
+ {
+ GParamSpecDouble *pspec = G_PARAM_SPEC_DOUBLE (spec);
+
+ lower = describe_double_constant (pspec->minimum);
+ upper = describe_double_constant (pspec->maximum);
+ if (GTKDOC_COMPARE_FLOAT (pspec->minimum, -G_MAXDOUBLE))
+ {
+ if (GTKDOC_COMPARE_FLOAT (pspec->maximum, G_MAXDOUBLE))
+ desc = g_strdup ("");
+ else
+ desc = g_strdup_printf ("<= %s", upper);
+ }
+ else if (GTKDOC_COMPARE_FLOAT (pspec->maximum, G_MAXDOUBLE))
+ desc = g_strdup_printf (">= %s", lower);
+ else
+ desc = g_strdup_printf ("[%s,%s]", lower, upper);
+ g_free (lower);
+ g_free (upper);
+ }
+#if GLIB_CHECK_VERSION (2, 12, 0)
+ else if (G_IS_PARAM_SPEC_GTYPE (spec))
+ {
+ GParamSpecGType *pspec = G_PARAM_SPEC_GTYPE (spec);
+ gboolean is_pointer;
+
+ desc = g_strdup (get_type_name (pspec->is_a_type, &is_pointer));
+ }
+#endif
+#if GLIB_CHECK_VERSION (2, 25, 9)
+ else if (G_IS_PARAM_SPEC_VARIANT (spec))
+ {
+ GParamSpecVariant *pspec = G_PARAM_SPEC_VARIANT (spec);
+ gchar *variant_type;
+
+ variant_type = g_variant_type_dup_string (pspec->type);
+ desc = g_strdup_printf ("GVariant<%s>", variant_type);
+ g_free (variant_type);
+ }
+#endif
+ else
+ {
+ desc = g_strdup ("");
+ }
+
+ return desc;
+}
+
+static gchar*
+describe_default (GParamSpec *spec)
+{
+ gchar *desc;
+
+ if (G_IS_PARAM_SPEC_CHAR (spec))
+ {
+ GParamSpecChar *pspec = G_PARAM_SPEC_CHAR (spec);
+
+ desc = g_strdup_printf ("%d", pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_UCHAR (spec))
+ {
+ GParamSpecUChar *pspec = G_PARAM_SPEC_UCHAR (spec);
+
+ desc = g_strdup_printf ("%u", pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_BOOLEAN (spec))
+ {
+ GParamSpecBoolean *pspec = G_PARAM_SPEC_BOOLEAN (spec);
+
+ desc = g_strdup_printf ("%s", pspec->default_value ? "TRUE" : "FALSE");
+ }
+ else if (G_IS_PARAM_SPEC_INT (spec))
+ {
+ GParamSpecInt *pspec = G_PARAM_SPEC_INT (spec);
+
+ desc = g_strdup_printf ("%d", pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_UINT (spec))
+ {
+ GParamSpecUInt *pspec = G_PARAM_SPEC_UINT (spec);
+
+ desc = g_strdup_printf ("%u", pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_LONG (spec))
+ {
+ GParamSpecLong *pspec = G_PARAM_SPEC_LONG (spec);
+
+ desc = g_strdup_printf ("%ld", pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_LONG (spec))
+ {
+ GParamSpecULong *pspec = G_PARAM_SPEC_ULONG (spec);
+
+ desc = g_strdup_printf ("%lu", pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_INT64 (spec))
+ {
+ GParamSpecInt64 *pspec = G_PARAM_SPEC_INT64 (spec);
+
+ desc = g_strdup_printf ("%" G_GINT64_FORMAT, pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_UINT64 (spec))
+ {
+ GParamSpecUInt64 *pspec = G_PARAM_SPEC_UINT64 (spec);
+
+ desc = g_strdup_printf ("%" G_GUINT64_FORMAT, pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_UNICHAR (spec))
+ {
+ GParamSpecUnichar *pspec = G_PARAM_SPEC_UNICHAR (spec);
+
+ if (g_unichar_isprint (pspec->default_value))
+ desc = g_strdup_printf ("'%c'", pspec->default_value);
+ else
+ desc = g_strdup_printf ("%u", pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_ENUM (spec))
+ {
+ GParamSpecEnum *pspec = G_PARAM_SPEC_ENUM (spec);
+
+ GEnumValue *value = g_enum_get_value (pspec->enum_class, pspec->default_value);
+ if (value)
+ desc = g_strdup_printf ("%s", value->value_name);
+ else
+ desc = g_strdup_printf ("%d", pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_FLAGS (spec))
+ {
+ GParamSpecFlags *pspec = G_PARAM_SPEC_FLAGS (spec);
+ guint default_value;
+ GString *acc;
+
+ default_value = pspec->default_value;
+ acc = g_string_new ("");
+
+ while (default_value)
+ {
+ GFlagsValue *value = g_flags_get_first_value (pspec->flags_class, default_value);
+
+ if (!value)
+ break;
+
+ if (acc->len > 0)
+ g_string_append (acc, "|");
+ g_string_append (acc, value->value_name);
+
+ default_value &= ~value->value;
+ }
+
+ if (default_value == 0)
+ desc = g_string_free (acc, FALSE);
+ else
+ {
+ desc = g_strdup_printf ("%d", pspec->default_value);
+ g_string_free (acc, TRUE);
+ }
+ }
+ else if (G_IS_PARAM_SPEC_FLOAT (spec))
+ {
+ GParamSpecFloat *pspec = G_PARAM_SPEC_FLOAT (spec);
+
+ /* make sure floats are output with a decimal dot irrespective of
+ * current locale. Use formatd since we want human-readable numbers
+ * and do not need the exact same bit representation when deserialising */
+ desc = g_malloc0 (G_ASCII_DTOSTR_BUF_SIZE);
+ g_ascii_formatd (desc, G_ASCII_DTOSTR_BUF_SIZE, "%g",
+ pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_DOUBLE (spec))
+ {
+ GParamSpecDouble *pspec = G_PARAM_SPEC_DOUBLE (spec);
+
+ /* make sure floats are output with a decimal dot irrespective of
+ * current locale. Use formatd since we want human-readable numbers
+ * and do not need the exact same bit representation when deserialising */
+ desc = g_malloc0 (G_ASCII_DTOSTR_BUF_SIZE);
+ g_ascii_formatd (desc, G_ASCII_DTOSTR_BUF_SIZE, "%g",
+ pspec->default_value);
+ }
+ else if (G_IS_PARAM_SPEC_STRING (spec))
+ {
+ GParamSpecString *pspec = G_PARAM_SPEC_STRING (spec);
+
+ if (pspec->default_value)
+ {
+ gchar *esc = g_strescape (pspec->default_value, NULL);
+
+ desc = g_strdup_printf ("\\"%s\\"", esc);
+
+ g_free (esc);
+ }
+ else
+ desc = g_strdup_printf ("NULL");
+ }
+ else
+ {
+ desc = g_strdup ("");
+ }
+
+ return desc;
+}
+
+
+static void
+output_object_args (FILE *fp, GType object_type)
+{
+ gpointer class;
+ const gchar *object_class_name;
+ guint arg;
+ gchar flags[16], *pos;
+ GParamSpec **properties;
+ guint n_properties;
+ gboolean child_prop;
+ gboolean style_prop;
+ gboolean is_pointer;
+ const gchar *type_name;
+ gchar *type_desc;
+ gchar *default_value;
+
+ if (G_TYPE_IS_OBJECT (object_type))
+ {
+ class = g_type_class_peek (object_type);
+ if (!class)
+ return;
+
+ properties = g_object_class_list_properties (class, &n_properties);
+ }
+#if GLIB_MAJOR_VERSION > 2 || (GLIB_MAJOR_VERSION == 2 && GLIB_MINOR_VERSION >= 3)
+ else if (G_TYPE_IS_INTERFACE (object_type))
+ {
+ class = g_type_default_interface_ref (object_type);
+
+ if (!class)
+ return;
+
+ properties = g_object_interface_list_properties (class, &n_properties);
+ }
+#endif
+ else
+ return;
+
+ object_class_name = g_type_name (object_type);
+
+ child_prop = FALSE;
+ style_prop = FALSE;
+
+ while (TRUE) {
+ qsort (properties, n_properties, sizeof (GParamSpec *), compare_param_specs);
+ for (arg = 0; arg < n_properties; arg++)
+ {
+ GParamSpec *spec = properties[arg];
+ const gchar *nick, *blurb, *dot;
+
+ if (spec->owner_type != object_type)
+ continue;
+
+ pos = flags;
+ /* We use one-character flags for simplicity. */
+ if (child_prop && !style_prop)
+ *pos++ = 'c';
+ if (style_prop)
+ *pos++ = 's';
+ if (spec->flags & G_PARAM_READABLE)
+ *pos++ = 'r';
+ if (spec->flags & G_PARAM_WRITABLE)
+ *pos++ = 'w';
+ if (spec->flags & G_PARAM_CONSTRUCT)
+ *pos++ = 'x';
+ if (spec->flags & G_PARAM_CONSTRUCT_ONLY)
+ *pos++ = 'X';
+ *pos = 0;
+
+ nick = g_param_spec_get_nick (spec);
+ blurb = g_param_spec_get_blurb (spec);
+
+ dot = "";
+ if (blurb) {
+ int str_len = strlen (blurb);
+ if (str_len > 0 && blurb[str_len - 1] != '.')
+ dot = ".";
+ }
+
+ type_desc = describe_type (spec);
+ default_value = describe_default (spec);
+ type_name = get_type_name (spec->value_type, &is_pointer);
+ fprintf (fp, "<ARG>\\n<NAME>%s::%s</NAME>\\n<TYPE>%s%s</TYPE>\\n<RANGE>%s</RANGE>\\n<FLAGS>%s</FLAGS>\\n<NICK>%s</NICK>\\n<BLURB>%s%s</BLURB>\\n<DEFAULT>%s</DEFAULT>\\n</ARG>\\n\\n",
+ object_class_name, g_param_spec_get_name (spec), type_name, is_pointer ? "*" : "", type_desc, flags, nick ? nick : "(null)", blurb ? blurb : "(null)", dot, default_value);
+ g_free (type_desc);
+ g_free (default_value);
+ }
+
+ g_free (properties);
+
+#ifdef GTK_IS_CONTAINER_CLASS
+ if (!child_prop && GTK_IS_CONTAINER_CLASS (class)) {
+ properties = gtk_container_class_list_child_properties (class, &n_properties);
+ child_prop = TRUE;
+ continue;
+ }
+#endif
+
+#ifdef GTK_IS_CELL_AREA_CLASS
+ if (!child_prop && GTK_IS_CELL_AREA_CLASS (class)) {
+ properties = gtk_cell_area_class_list_cell_properties (class, &n_properties);
+ child_prop = TRUE;
+ continue;
+ }
+#endif
+
+#ifdef GTK_IS_WIDGET_CLASS
+#if GTK_CHECK_VERSION(2,1,0)
+ if (!style_prop && GTK_IS_WIDGET_CLASS (class)) {
+ properties = gtk_widget_class_list_style_properties (GTK_WIDGET_CLASS (class), &n_properties);
+ style_prop = TRUE;
+ continue;
+ }
+#endif
+#endif
+
+ break;
+ }
+}
+EOT
+
+close OUTPUT;
+
+# Compile and run our file
+
+$CC = $ENV{CC} ? $ENV{CC} : "gcc";
+$LD = $ENV{LD} ? $ENV{LD} : $CC;
+$CFLAGS = $ENV{CFLAGS} ? "$ENV{CFLAGS}" : "";
+$LDFLAGS = $ENV{LDFLAGS} ? $ENV{LDFLAGS} : "";
+
+my $o_file;
+if ($CC =~ /libtool/) {
+ $o_file = "$MODULE-scan.lo"
+} else {
+ $o_file = "$MODULE-scan.o"
+}
+
+my $stdout="";
+if (!defined($VERBOSE) or $VERBOSE eq "0") {
+ $stdout=">/dev/null";
+}
+
+# Compiling scanner
+$command = "$CC $stdout $CFLAGS -c -o $o_file $MODULE-scan.c";
+system("($command)") == 0 or die "Compilation of scanner failed: $!\n";
+
+# Linking scanner
+$command = "$LD $stdout -o $MODULE-scan $o_file $LDFLAGS";
+system($command) == 0 or die "Linking of scanner failed: $!\n";
+
+# Running scanner $MODULE-scan ";
+system("sh -c ./$MODULE-scan") == 0 or die "Scan failed: $!\n";
+
+if (!defined($ENV{"GTK_DOC_KEEP_INTERMEDIATE"})) {
+ unlink "./$MODULE-scan.c", "./$MODULE-scan.o", "./$MODULE-scan.lo", "./$MODULE-scan";
+}
+
+&UpdateFileIfChanged ($old_hierarchy_filename, $new_hierarchy_filename, 0);
+# we will merge these in scangobj-merge.py
+#&UpdateFileIfChanged ($old_interfaces_filename, $new_interfaces_filename, 0);
+#&UpdateFileIfChanged ($old_prerequisites_filename, $new_prerequisites_filename, 0);
+#&UpdateFileIfChanged ($old_signals_filename, $new_signals_filename, 0);
+#&UpdateFileIfChanged ($old_args_filename, $new_args_filename, 0);
+
--- /dev/null
+# This is an include file specifically tuned for building documentation
+# for GStreamer plug-ins
+
+help:
+ @echo
+ @echo "If you are a doc maintainer, run 'make update' to update"
+ @echo "the documentation files maintained in git"
+ @echo
+ @echo Other useful make targets:
+ @echo
+ @echo check-inspected-versions: make sure the inspected plugin info
+ @echo is up to date before a release
+ @echo
+
+# update the stuff maintained by doc maintainers
+update: scanobj-update
+ $(MAKE) check-outdated-docs
+
+# We set GPATH here; this gives us semantics for GNU make
+# which are more like other make's VPATH, when it comes to
+# whether a source that is a target of one rule is then
+# searched for in VPATH/GPATH.
+#
+GPATH = $(srcdir)
+
+# thomas: make docs parallel installable
+TARGET_DIR=$(HTML_DIR)/$(DOC_MODULE)-@GST_API_VERSION@
+
+MAINTAINER_DOC_STAMPS = \
+ scanobj-build.stamp
+
+EXTRA_DIST = \
+ $(MAINTAINER_DOC_STAMPS) \
+ $(srcdir)/inspect/*.xml \
+ $(SCANOBJ_FILES) \
+ $(content_files) \
+ $(extra_files) \
+ $(HTML_IMAGES) \
+ $(DOC_MAIN_SGML_FILE) \
+ $(DOC_OVERRIDES) \
+ $(DOC_MODULE)-sections.txt
+
+# we don't add scanobj-build.stamp here since they are built manually by docs
+# maintainers and result is commited to git
+DOC_STAMPS = \
+ scan-build.stamp \
+ tmpl-build.stamp \
+ sgml-build.stamp \
+ html-build.stamp \
+ scan.stamp \
+ tmpl.stamp \
+ sgml.stamp \
+ html.stamp
+
+# files generated/updated by gtkdoc-scangobj
+SCANOBJ_FILES = \
+ $(DOC_MODULE).args \
+ $(DOC_MODULE).hierarchy \
+ $(DOC_MODULE).interfaces \
+ $(DOC_MODULE).prerequisites \
+ $(DOC_MODULE).signals \
+ $(DOC_MODULE).types
+
+SCANOBJ_FILES_O = \
+ .libs/$(DOC_MODULE)-scan.o
+
+# files generated/updated by gtkdoc-scan
+SCAN_FILES = \
+ $(DOC_MODULE)-sections.txt \
+ $(DOC_MODULE)-overrides.txt \
+ $(DOC_MODULE)-decl.txt \
+ $(DOC_MODULE)-decl-list.txt
+
+
+REPORT_FILES = \
+ $(DOC_MODULE)-undocumented.txt \
+ $(DOC_MODULE)-undeclared.txt \
+ $(DOC_MODULE)-unused.txt
+
+CLEANFILES = \
+ $(SCANOBJ_FILES_O) \
+ $(REPORT_FILES) \
+ $(DOC_STAMPS) \
+ inspect-registry.xml
+
+INSPECT_DIR = inspect
+
+if ENABLE_GTK_DOC
+all-local: html-build.stamp
+
+### inspect GStreamer plug-ins; done by documentation maintainer ###
+
+# only look at the plugins in this module when building inspect .xml stuff
+INSPECT_REGISTRY=$(top_builddir)/docs/plugins/inspect-registry.xml
+INSPECT_ENVIRONMENT=\
+ LC_ALL=C \
+ GST_PLUGIN_SYSTEM_PATH_1_0= \
+ GST_PLUGIN_PATH_1_0=$(top_builddir)/gst:$(top_builddir)/sys:$(top_builddir)/ext:$(top_builddir)/plugins:$(top_builddir)/src:$(top_builddir)/gnl \
+ GST_REGISTRY_1_0=$(INSPECT_REGISTRY) \
+ PKG_CONFIG_PATH="$(GST_PKG_CONFIG_PATH)" \
+ $(INSPECT_EXTRA_ENVIRONMENT)
+
+#### scan gobjects; done by documentation maintainer ####
+scanobj-update:
+ -rm scanobj-build.stamp
+ $(MAKE) scanobj-build.stamp
+
+# gstdoc-scanobj produces 5 output files (.new)
+# scangobj-merge.py merges them into the file which we commit later
+# TODO: also merge the hierarchy
+scanobj-build.stamp: $(SCANOBJ_DEPS) $(basefiles)
+ @echo " DOC Introspecting gobjects"
+ @if test x"$(srcdir)" != x. ; then \
+ for f in $(SCANOBJ_FILES) $(SCAN_FILES); \
+ do \
+ if test -e $(srcdir)/$$f; then cp -u $(srcdir)/$$f . ; fi; \
+ done; \
+ fi; \
+ mkdir -p $(INSPECT_DIR); \
+ scanobj_options=""; \
+ if test "x$(V)" = "x1"; then \
+ scanobj_options="--verbose"; \
+ fi; \
+ $(INSPECT_ENVIRONMENT) \
+ CC="$(GTKDOC_CC)" LD="$(GTKDOC_LD)" \
+ CFLAGS="$(GTKDOC_CFLAGS) $(CFLAGS) $(WARNING_CFLAGS)" \
+ LDFLAGS="$(GTKDOC_LIBS) $(LDFLAGS)" \
+ $(GST_DOC_SCANOBJ) $$scanobj_options --type-init-func="gst_init(NULL,NULL)" \
+ --module=$(DOC_MODULE) --source=$(PACKAGE) --inspect-dir=$(INSPECT_DIR) && \
+ echo " DOC Merging introspection data" && \
+ $(PYTHON) \
+ $(top_srcdir)/common/scangobj-merge.py $(DOC_MODULE) || exit 1; \
+ if test x"$(srcdir)" != x. ; then \
+ for f in $(SCANOBJ_FILES); \
+ do \
+ cmp -s ./$$f $(srcdir)/$$f || cp ./$$f $(srcdir)/ ; \
+ done; \
+ fi; \
+ touch scanobj-build.stamp
+
+$(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(SCANOBJ_FILES_O): scan-build.stamp
+ @true
+
+### scan headers; done on every build ###
+scan-build.stamp: $(HFILE_GLOB) $(EXTRA_HFILES) $(basefiles) scanobj-build.stamp
+ @echo ' DOC Scanning header files'
+ @if test x"$(srcdir)" != x. ; then \
+ for f in $(SCANOBJ_FILES) $(SCAN_FILES); \
+ do \
+ if test -e $(srcdir)/$$f; then cp -u $(srcdir)/$$f . ; fi; \
+ done; \
+ fi
+ @_source_dir='' ; \
+ for i in $(DOC_SOURCE_DIR) ; do \
+ _source_dir="$${_source_dir} --source-dir=$$i" ; \
+ done ; \
+ gtkdoc-scan \
+ $(SCAN_OPTIONS) $(EXTRA_HFILES) \
+ --module=$(DOC_MODULE) \
+ $${_source_dir} \
+ --ignore-headers="$(IGNORE_HFILES)"; \
+ touch scan-build.stamp
+
+#### update templates; done on every build ####
+
+# in a non-srcdir build, we need to copy files from the previous step
+# and the files from previous runs of this step
+tmpl-build.stamp: $(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(DOC_MODULE)-sections.txt $(DOC_OVERRIDES)
+ @echo ' DOC Rebuilding template files'
+ @if test x"$(srcdir)" != x. ; then \
+ for f in $(SCANOBJ_FILES) $(SCAN_FILES); \
+ do \
+ if test -e $(srcdir)/$$f; then cp -u $(srcdir)/$$f . ; fi; \
+ done; \
+ fi
+ @gtkdoc-mktmpl --module=$(DOC_MODULE)
+ @$(PYTHON) \
+ $(top_srcdir)/common/mangle-tmpl.py $(srcdir)/$(INSPECT_DIR) tmpl
+ @touch tmpl-build.stamp
+
+tmpl.stamp: tmpl-build.stamp
+ @true
+
+#### xml ####
+
+sgml-build.stamp: tmpl.stamp scan-build.stamp $(CFILE_GLOB) $(top_srcdir)/common/plugins.xsl $(expand_content_files)
+ @echo ' DOC Building XML'
+ @-mkdir -p xml
+ @for a in $(srcdir)/$(INSPECT_DIR)/*.xml; do \
+ xsltproc --stringparam module $(MODULE) \
+ $(top_srcdir)/common/plugins.xsl $$a > xml/`basename $$a`; done
+ @for f in $(EXAMPLE_CFILES); do \
+ $(PYTHON) $(top_srcdir)/common/c-to-xml.py $$f > xml/element-`basename $$f .c`.xml; done
+ @gtkdoc-mkdb \
+ --module=$(DOC_MODULE) \
+ --source-dir=$(DOC_SOURCE_DIR) \
+ --expand-content-files="$(expand_content_files)" \
+ --main-sgml-file=$(srcdir)/$(DOC_MAIN_SGML_FILE) \
+ --output-format=xml \
+ --ignore-files="$(IGNORE_HFILES) $(IGNORE_CFILES)" \
+ $(MKDB_OPTIONS)
+ @cp ../version.entities xml
+ @touch sgml-build.stamp
+
+sgml.stamp: sgml-build.stamp
+ @true
+
+#### html ####
+
+html-build.stamp: sgml.stamp $(DOC_MAIN_SGML_FILE) $(content_files)
+ @echo ' DOC Building HTML'
+ @rm -rf html
+ @mkdir html
+ @cp $(srcdir)/$(DOC_MAIN_SGML_FILE) html
+ @for f in $(content_files); do cp $(srcdir)/$$f html; done
+ @cp -pr xml html
+ @cp ../version.entities html
+ @mkhtml_options=""; \
+ gtkdoc-mkhtml 2>&1 --help | grep >/dev/null "\-\-verbose"; \
+ if test "$(?)" = "0"; then \
+ if test "x$(V)" = "x1"; then \
+ mkhtml_options="$$mkhtml_options --verbose"; \
+ fi; \
+ fi; \
+ cd html && gtkdoc-mkhtml $$mkhtml_options $(DOC_MODULE)-@GST_API_VERSION@ $(DOC_MAIN_SGML_FILE)
+ @rm -f html/$(DOC_MAIN_SGML_FILE)
+ @rm -rf html/xml
+ @rm -f html/version.entities
+ @test "x$(HTML_IMAGES)" = "x" || for i in "" $(HTML_IMAGES) ; do \
+ if test "$$i" != ""; then cp $(srcdir)/$$i html ; fi; done
+ @echo ' DOC Fixing cross-references'
+ @gtkdoc-fixxref --module=$(DOC_MODULE) --module-dir=html --html-dir=$(HTML_DIR) $(FIXXREF_OPTIONS)
+ @touch html-build.stamp
+
+clean-local-gtkdoc:
+ @rm -rf xml tmpl html
+# clean files copied for nonsrcdir templates build
+ @if test x"$(srcdir)" != x. ; then \
+ rm -rf $(SCANOBJ_FILES) $(SCAN_FILES) $(REPORT_FILES) \
+ $(MAINTAINER_DOC_STAMPS); \
+ fi
+else
+all-local:
+clean-local-gtkdoc:
+endif
+
+clean-local: clean-local-gtkdoc
+ @rm -f *~ *.bak
+ @rm -rf .libs
+
+distclean-local:
+ @rm -f $(REPORT_FILES) \
+ $(DOC_MODULE)-decl-list.txt $(DOC_MODULE)-decl.txt
+ @rm -rf tmpl/*.sgml.bak
+ @rm -f $(DOC_MODULE).hierarchy
+ @rm -f *.stamp || true
+ @if test "$(abs_srcdir)" != "$(abs_builddir)" ; then \
+ rm -f $(DOC_MODULE)-docs.sgml ; \
+ rm -f $(DOC_MODULE).types ; \
+ rm -f $(DOC_MODULE).interfaces ; \
+ rm -f $(DOC_MODULE)-overrides.txt ; \
+ rm -f $(DOC_MODULE).prerequisites ; \
+ rm -f $(DOC_MODULE)-sections.txt ; \
+ rm -rf tmpl/*.sgml ; \
+ rm -rf $(INSPECT_DIR); \
+ fi
+ @rm -rf *.o
+
+MAINTAINERCLEANFILES = $(MAINTAINER_DOC_STAMPS)
+
+# thomas: make docs parallel installable; devhelp requires majorminor too
+install-data-local:
+ (installfiles=`echo $(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css`; \
+ if test "$$installfiles" = '$(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css'; \
+ then echo '-- Nothing to install' ; \
+ else \
+ $(mkinstalldirs) $(DESTDIR)$(TARGET_DIR); \
+ for i in $$installfiles; do \
+ echo '-- Installing '$$i ; \
+ $(INSTALL_DATA) $$i $(DESTDIR)$(TARGET_DIR); \
+ done; \
+ pngfiles=`echo ./html/*.png`; \
+ if test "$$pngfiles" != './html/*.png'; then \
+ for i in $$pngfiles; do \
+ echo '-- Installing '$$i ; \
+ $(INSTALL_DATA) $$i $(DESTDIR)$(TARGET_DIR); \
+ done; \
+ fi; \
+ echo '-- Installing $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2' ; \
+ if test -e $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; then \
+ $(INSTALL_DATA) $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2 \
+ $(DESTDIR)$(TARGET_DIR)/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; \
+ fi; \
+ $(GTKDOC_REBASE) --relative --dest-dir=$(DESTDIR) --html-dir=$(DESTDIR)$(TARGET_DIR) || true ; \
+ fi)
+uninstall-local:
+ if test -d $(DESTDIR)$(TARGET_DIR); then \
+ rm -rf $(DESTDIR)$(TARGET_DIR)/*; \
+ rmdir -p $(DESTDIR)$(TARGET_DIR) 2>/dev/null || true; \
+ else \
+ echo '-- Nothing to uninstall' ; \
+ fi;
+
+#
+# Checks
+#
+if ENABLE_GTK_DOC
+check-hierarchy: $(DOC_MODULE).hierarchy
+ @if grep ' ' $(DOC_MODULE).hierarchy; then \
+ echo "$(DOC_MODULE).hierarchy contains tabs, please fix"; \
+ /bin/false; \
+ fi
+
+check: check-hierarchy
+endif
+
+# wildcard is apparently not portable to other makes, hence the use of find
+inspect_files = $(shell find $(srcdir)/$(INSPECT_DIR) -name '*.xml')
+
+check-inspected-versions:
+ @echo Checking plugin versions of inspected plugin data ...; \
+ fail=0 ; \
+ for each in $(inspect_files) ; do \
+ if (grep -H '<version>' $$each | grep -v '<version>$(VERSION)'); then \
+ echo $$each should be fixed to say version $(VERSION) or be removed ; \
+ echo "sed -i -e 's/<version.*version>/<version>$(VERSION)<\/version>/'" $$each; \
+ echo ; \
+ fail=1; \
+ fi ; \
+ done ; \
+ exit $$fail
+
+check-outdated-docs:
+ $(AM_V_GEN)echo Checking for outdated plugin inspect data ...; \
+ fail=0 ; \
+ if [ -d $(top_srcdir)/.git/ ]; then \
+ files=`find $(srcdir)/inspect/ -name '*xml'`; \
+ for f in $$files; do \
+ ver=`grep '<version>$(PACKAGE_VERSION)</version>' $$f`; \
+ if test "x$$ver" = "x"; then \
+ plugin=`echo $$f | sed -e 's/^.*plugin-//' -e 's/.xml//'`; \
+ # echo "Checking $$plugin $$f"; \
+ pushd "$(top_srcdir)" >/dev/null; \
+ pinit=`git grep -A3 GST_PLUGIN_DEFINE -- ext/ gst/ sys/ | grep "\"$$plugin\""`; \
+ popd >/dev/null; \
+ # echo "[$$pinit]"; \
+ if test "x$$pinit" = "x"; then \
+ printf " **** outdated docs for plugin %-15s: %s\n" $$plugin $$f; \
+ fail=1; \
+ fi; \
+ fi; \
+ done; \
+ fi ; \
+ exit $$fail
+
+#
+# Require gtk-doc when making dist
+#
+if ENABLE_GTK_DOC
+dist-check-gtkdoc:
+else
+dist-check-gtkdoc:
+ @echo "*** gtk-doc must be installed and enabled in order to make dist"
+ @false
+endif
+
+# FIXME: decide whether we want to dist generated html or not
+# also this only works, if the project has been build before
+# we could dist html only if its there, but that might lead to missing html in
+# tarballs
+dist-hook: dist-check-gtkdoc dist-hook-local
+ mkdir $(distdir)/html
+ cp html/* $(distdir)/html
+ -cp $(srcdir)/$(DOC_MODULE).types $(distdir)/
+ -cp $(srcdir)/$(DOC_MODULE)-sections.txt $(distdir)/
+ cd $(distdir) && rm -f $(DISTCLEANFILES)
+ -gtkdoc-rebase --online --relative --html-dir=$(distdir)/html
+
+.PHONY : dist-hook-local docs check-outdated-docs inspect
+
+# avoid spurious build errors when distchecking with -jN
+.NOTPARALLEL:
--- /dev/null
+###########################################################################
+# Everything below here is generic and you shouldn't need to change it.
+###########################################################################
+# thomas: except of course that we did
+
+# thomas: copied from glib-2
+# We set GPATH here; this gives us semantics for GNU make
+# which are more like other make's VPATH, when it comes to
+# whether a source that is a target of one rule is then
+# searched for in VPATH/GPATH.
+#
+GPATH = $(srcdir)
+
+# thomas: make docs parallel installable
+TARGET_DIR=$(HTML_DIR)/$(DOC_MODULE)-@GST_API_VERSION@
+
+EXTRA_DIST = \
+ $(content_files) \
+ $(extra_files) \
+ $(HTML_IMAGES) \
+ $(DOC_MAIN_SGML_FILE) \
+ $(DOC_MODULE).types \
+ $(DOC_OVERRIDES) \
+ $(DOC_MODULE)-sections.txt
+
+DOC_STAMPS = \
+ setup-build.stamp \
+ scan-build.stamp \
+ sgml-build.stamp \
+ html-build.stamp \
+ sgml.stamp \
+ html.stamp
+
+SCANOBJ_FILES = \
+ $(DOC_MODULE).args \
+ $(DOC_MODULE).hierarchy \
+ $(DOC_MODULE).interfaces \
+ $(DOC_MODULE).prerequisites \
+ $(DOC_MODULE).signals \
+ .libs/$(DOC_MODULE)-scan.o
+
+REPORT_FILES = \
+ $(DOC_MODULE)-undocumented.txt \
+ $(DOC_MODULE)-undeclared.txt \
+ $(DOC_MODULE)-unused.txt
+
+CLEANFILES = $(SCANOBJ_FILES) $(REPORT_FILES) $(DOC_STAMPS) doc-registry.xml
+
+if ENABLE_GTK_DOC
+all-local: html-build.stamp
+
+#### setup ####
+
+setup-build.stamp: $(content_files)
+ -@if test "$(abs_srcdir)" != "$(abs_builddir)" ; then \
+ echo ' DOC Preparing build'; \
+ files=`echo $(DOC_MAIN_SGML_FILE) $(DOC_OVERRIDES) $(DOC_MODULE)-sections.txt $(DOC_MODULE).types $(content_files)`; \
+ if test "x$$files" != "x" ; then \
+ for file in $$files ; do \
+ test -f $(abs_srcdir)/$$file && \
+ cp -pu $(abs_srcdir)/$$file $(abs_builddir)/ || true; \
+ done; \
+ fi; \
+ fi
+ @touch setup-build.stamp
+
+#### scan ####
+
+# in the case of non-srcdir builds, the built gst directory gets added
+# to gtk-doc scanning; but only then, to avoid duplicates
+scan-build.stamp: $(HFILE_GLOB) $(CFILE_GLOB)
+ @echo ' DOC Scanning header files'
+ @_source_dir='' ; \
+ for i in $(DOC_SOURCE_DIR) ; do \
+ _source_dir="$${_source_dir} --source-dir=$$i" ; \
+ done ; \
+ gtkdoc-scan \
+ $(SCAN_OPTIONS) $(EXTRA_HFILES) \
+ --module=$(DOC_MODULE) \
+ $${_source_dir} \
+ --ignore-headers="$(IGNORE_HFILES)"
+ @if grep -l '^..*$$' $(DOC_MODULE).types > /dev/null; then \
+ echo " DOC Introspecting gobjects"; \
+ GST_PLUGIN_SYSTEM_PATH_1_0=`cd $(top_builddir) && pwd` \
+ GST_PLUGIN_PATH_1_0= \
+ GST_REGISTRY_1_0=doc-registry.xml \
+ $(GTKDOC_EXTRA_ENVIRONMENT) \
+ CC="$(GTKDOC_CC)" LD="$(GTKDOC_LD)" \
+ CFLAGS="$(GTKDOC_CFLAGS) $(CFLAGS)" \
+ LDFLAGS="$(GTKDOC_LIBS) $(LDFLAGS)" \
+ gtkdoc-scangobj --type-init-func="gst_init(NULL,NULL)" \
+ --module=$(DOC_MODULE) ; \
+ else \
+ for i in $(SCANOBJ_FILES) ; do \
+ test -f $$i || touch $$i ; \
+ done \
+ fi
+ @touch scan-build.stamp
+
+$(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(DOC_MODULE)-sections.txt $(DOC_MODULE)-overrides.txt: scan-build.stamp
+ @true
+
+#### xml ####
+
+sgml-build.stamp: setup-build.stamp $(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(DOC_MODULE)-sections.txt $(expand_content_files)
+ @echo ' DOC Building XML'
+ @gtkdoc-mkdb --module=$(DOC_MODULE) --source-dir=$(DOC_SOURCE_DIR) --expand-content-files="$(expand_content_files)" --main-sgml-file=$(DOC_MAIN_SGML_FILE) --output-format=xml $(MKDB_OPTIONS)
+ @cp ../version.entities xml
+ @touch sgml-build.stamp
+
+sgml.stamp: sgml-build.stamp
+ @true
+
+#### html ####
+
+html-build.stamp: sgml.stamp $(DOC_MAIN_SGML_FILE) $(content_files)
+ @echo ' DOC Building HTML'
+ @rm -rf html
+ @mkdir html
+ @cp -pr xml html
+ @cp ../version.entities ./
+ @mkhtml_options=""; \
+ gtkdoc-mkhtml 2>&1 --help | grep >/dev/null "\-\-verbose"; \
+ if test "$(?)" = "0"; then \
+ if test "x$(V)" = "x1"; then \
+ mkhtml_options="$$mkhtml_options --verbose"; \
+ fi; \
+ fi; \
+ @gtkdoc-mkhtml 2>&1 --help | grep >/dev/null "\-\-path"; \
+ if test "$(?)" = "0"; then \
+ mkhtml_options=--path="$(abs_srcdir)"; \
+ fi; \
+ cd html && gtkdoc-mkhtml $$mkhtml_options $(MKHTML_OPTIONS) $(DOC_MODULE)-@GST_API_VERSION@ ../$(DOC_MAIN_SGML_FILE)
+ @rm -rf html/xml
+ @rm -f version.entities
+ @test "x$(HTML_IMAGES)" = "x" || ( cd $(srcdir) && cp $(HTML_IMAGES) $(abs_builddir)/html )
+ @echo ' DOC Fixing cross-references'
+ @gtkdoc-fixxref --module=$(DOC_MODULE) --module-dir=html --html-dir=$(HTML_DIR) $(FIXXREF_OPTIONS)
+ @touch html-build.stamp
+
+clean-local-gtkdoc:
+ @rm -rf xml tmpl html
+# clean files copied for nonsrcdir templates build
+ @if test x"$(srcdir)" != x. ; then \
+ rm -rf $(DOC_MODULE).types; \
+ fi
+else
+all-local:
+clean-local-gtkdoc:
+endif
+
+clean-local: clean-local-gtkdoc
+ @rm -f *~ *.bak
+ @rm -rf .libs
+
+distclean-local:
+ @rm -f $(REPORT_FILES) \
+ $(DOC_MODULE)-decl-list.txt $(DOC_MODULE)-decl.txt
+ @rm -rf tmpl/*.sgml.bak
+ @rm -f $(DOC_MODULE).hierarchy
+ @rm -f *.stamp || true
+ @if test "$(abs_srcdir)" != "$(abs_builddir)" ; then \
+ rm -f $(DOC_MAIN_SGML_FILE) ; \
+ rm -f $(DOC_OVERRIDES) ; \
+ rm -f $(DOC_MODULE).types ; \
+ rm -f $(DOC_MODULE).interfaces ; \
+ rm -f $(DOC_MODULE).prerequisites ; \
+ rm -f $(DOC_MODULE)-sections.txt ; \
+ rm -f $(content_files) ; \
+ rm -rf tmpl/*.sgml ; \
+ fi
+ @rm -rf *.o
+
+maintainer-clean-local: clean
+ @cd $(srcdir) && rm -rf html \
+ xml $(DOC_MODULE)-decl-list.txt $(DOC_MODULE)-decl.txt
+
+# thomas: make docs parallel installable; devhelp requires majorminor too
+install-data-local:
+ (installfiles=`echo $(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css`; \
+ if test "$$installfiles" = '$(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css'; \
+ then echo '-- Nothing to install' ; \
+ else \
+ $(mkinstalldirs) $(DESTDIR)$(TARGET_DIR); \
+ for i in $$installfiles; do \
+ echo '-- Installing '$$i ; \
+ $(INSTALL_DATA) $$i $(DESTDIR)$(TARGET_DIR); \
+ done; \
+ echo '-- Installing $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2' ; \
+ if test -e $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; then \
+ $(INSTALL_DATA) $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2 \
+ $(DESTDIR)$(TARGET_DIR)/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; \
+ fi; \
+ $(GTKDOC_REBASE) --relative --dest-dir=$(DESTDIR) --html-dir=$(DESTDIR)$(TARGET_DIR) || true ; \
+ fi)
+uninstall-local:
+ if test -d $(DESTDIR)$(TARGET_DIR); then \
+ rm -rf $(DESTDIR)$(TARGET_DIR)/*; \
+ rmdir -p $(DESTDIR)$(TARGET_DIR) 2>/dev/null || true; \
+ else \
+ echo '-- Nothing to uninstall' ; \
+ fi;
+
+
+#
+# Require gtk-doc when making dist
+#
+if ENABLE_GTK_DOC
+dist-check-gtkdoc:
+else
+dist-check-gtkdoc:
+ @echo "*** gtk-doc must be installed and enabled in order to make dist"
+ @false
+endif
+
+dist-hook: dist-check-gtkdoc dist-hook-local
+ mkdir $(distdir)/html
+ cp html/* $(distdir)/html
+ -cp $(srcdir)/$(DOC_MODULE).types $(distdir)/
+ -cp $(srcdir)/$(DOC_MODULE)-sections.txt $(distdir)/
+ cd $(distdir) && rm -f $(DISTCLEANFILES)
+ -gtkdoc-rebase --online --relative --html-dir=$(distdir)/html
+
+.PHONY : dist-hook-local docs
+
+# avoid spurious build errors when distchecking with -jN
+.NOTPARALLEL:
--- /dev/null
+EXTRA_DIST = \
+ README \
+ as-ac-expand.m4 \
+ as-auto-alt.m4 \
+ as-compiler-flag.m4 \
+ as-compiler.m4 \
+ as-docbook.m4 \
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+ gst-args.m4 \
+ gst-check.m4 \
+ gst-debuginfo.m4 \
+ gst-default.m4 \
+ gst-doc.m4 \
+ gst-dowhile.m4 \
+ gst-error.m4 \
+ gst-feature.m4 \
+ gst-function.m4 \
+ gst-gettext.m4 \
+ gst-glib2.m4 \
+ gst-libxml2.m4 \
+ gst-parser.m4 \
+ gst-package-release-datetime.m4 \
+ gst-platform.m4 \
+ gst-plugindir.m4 \
+ gst-plugin-docs.m4 \
+ gst-valgrind.m4 \
+ gst-x11.m4 \
+ gst.m4 \
+ gtk-doc.m4 \
+ introspection.m4 \
+ pkg.m4 \
+ check.m4 \
+ orc.m4
--- /dev/null
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+ glib-gettext.m4 \
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+ gst-args.m4 \
+ gst-check.m4 \
+ gst-debuginfo.m4 \
+ gst-default.m4 \
+ gst-doc.m4 \
+ gst-dowhile.m4 \
+ gst-error.m4 \
+ gst-feature.m4 \
+ gst-function.m4 \
+ gst-gettext.m4 \
+ gst-glib2.m4 \
+ gst-libxml2.m4 \
+ gst-parser.m4 \
+ gst-package-release-datetime.m4 \
+ gst-platform.m4 \
+ gst-plugindir.m4 \
+ gst-plugin-docs.m4 \
+ gst-valgrind.m4 \
+ gst-x11.m4 \
+ gst.m4 \
+ gtk-doc.m4 \
+ introspection.m4 \
+ pkg.m4 \
+ check.m4 \
+ orc.m4
+
+all: all-am
+
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+ exit 1;; \
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+ $(am__cd) $(top_srcdir) && \
+ $(AUTOMAKE) --gnu common/m4/Makefile
+.PRECIOUS: Makefile
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+ cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
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+
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+
+
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+ list='$(DISTFILES)'; \
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+ -e "s|^$$topsrcdirstrip/|$(top_builddir)/|;t"`; \
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+ */*) $(MKDIR_P) `echo "$$dist_files" | \
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+ sort -u` ;; \
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+ fi; \
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+ cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
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+ else \
+ test -f "$(distdir)/$$file" \
+ || cp -p $$d/$$file "$(distdir)/$$file" \
+ || exit 1; \
+ fi; \
+ done
+check-am: all-am
+check: check-am
+all-am: Makefile
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+ install; \
+ else \
+ $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+ install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+ "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+ fi
+mostlyclean-generic:
+
+clean-generic:
+
+distclean-generic:
+ -test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+ -test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+ @echo "This command is intended for maintainers to use"
+ @echo "it deletes files that may require special tools to rebuild."
+clean: clean-am
+
+clean-am: clean-generic clean-libtool mostlyclean-am
+
+distclean: distclean-am
+ -rm -f Makefile
+distclean-am: clean-am distclean-generic
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+ -rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-generic mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
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+ps: ps-am
+
+ps-am:
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+uninstall-am:
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+.MAKE: install-am install-strip
+
+.PHONY: all all-am check check-am clean clean-generic clean-libtool \
+ cscopelist-am ctags-am distclean distclean-generic \
+ distclean-libtool distdir dvi dvi-am html html-am info info-am \
+ install install-am install-data install-data-am install-dvi \
+ install-dvi-am install-exec install-exec-am install-html \
+ install-html-am install-info install-info-am install-man \
+ install-pdf install-pdf-am install-ps install-ps-am \
+ install-strip installcheck installcheck-am installdirs \
+ maintainer-clean maintainer-clean-generic mostlyclean \
+ mostlyclean-generic mostlyclean-libtool pdf pdf-am ps ps-am \
+ tags-am uninstall uninstall-am
+
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
--- /dev/null
+All aclocal .m4 files we need are put here and cat'd to acinclude.m4 in
+the source root. Official ones (taken from the relevant devel packages)
+are named as-is, unofficial ones (or changed ones) get a gst-prefix.
--- /dev/null
+dnl as-ac-expand.m4 0.2.0
+dnl autostars m4 macro for expanding directories using configure's prefix
+dnl thomas@apestaart.org
+
+dnl AS_AC_EXPAND(VAR, CONFIGURE_VAR)
+dnl example
+dnl AS_AC_EXPAND(SYSCONFDIR, $sysconfdir)
+dnl will set SYSCONFDIR to /usr/local/etc if prefix=/usr/local
+
+AC_DEFUN([AS_AC_EXPAND],
+[
+ EXP_VAR=[$1]
+ FROM_VAR=[$2]
+
+ dnl first expand prefix and exec_prefix if necessary
+ prefix_save=$prefix
+ exec_prefix_save=$exec_prefix
+
+ dnl if no prefix given, then use /usr/local, the default prefix
+ if test "x$prefix" = "xNONE"; then
+ prefix="$ac_default_prefix"
+ fi
+ dnl if no exec_prefix given, then use prefix
+ if test "x$exec_prefix" = "xNONE"; then
+ exec_prefix=$prefix
+ fi
+
+ full_var="$FROM_VAR"
+ dnl loop until it doesn't change anymore
+ while true; do
+ new_full_var="`eval echo $full_var`"
+ if test "x$new_full_var" = "x$full_var"; then break; fi
+ full_var=$new_full_var
+ done
+
+ dnl clean up
+ full_var=$new_full_var
+ AC_SUBST([$1], "$full_var")
+
+ dnl restore prefix and exec_prefix
+ prefix=$prefix_save
+ exec_prefix=$exec_prefix_save
+])
--- /dev/null
+dnl as-auto-alt.m4 0.0.2
+dnl autostars m4 macro for supplying alternate autotools versions to configure
+dnl thomas@apestaart.org
+dnl
+dnl AS_AUTOTOOLS_ALTERNATE()
+dnl
+dnl supplies --with arguments for autoconf, autoheader, automake, aclocal
+
+AC_DEFUN([AS_AUTOTOOLS_ALTERNATE],
+[
+ dnl allow for different autoconf version
+ AC_ARG_WITH(autoconf,
+ AC_HELP_STRING([--with-autoconf],
+ [use a different autoconf for regeneration of Makefiles]),
+ [
+ unset AUTOCONF
+ AM_MISSING_PROG(AUTOCONF, ${withval})
+ AC_MSG_NOTICE([Using $AUTOCONF as autoconf])
+ ])
+
+ dnl allow for different autoheader version
+ AC_ARG_WITH(autoheader,
+ AC_HELP_STRING([--with-autoheader],
+ [use a different autoheader for regeneration of Makefiles]),
+ [
+ unset AUTOHEADER
+ AM_MISSING_PROG(AUTOHEADER, ${withval})
+ AC_MSG_NOTICE([Using $AUTOHEADER as autoheader])
+ ])
+
+ dnl allow for different automake version
+ AC_ARG_WITH(automake,
+ AC_HELP_STRING([--with-automake],
+ [use a different automake for regeneration of Makefiles]),
+ [
+ unset AUTOMAKE
+ AM_MISSING_PROG(AUTOMAKE, ${withval})
+ AC_MSG_NOTICE([Using $AUTOMAKE as automake])
+ ])
+
+ dnl allow for different aclocal version
+ AC_ARG_WITH(aclocal,
+ AC_HELP_STRING([--with-aclocal],
+ [use a different aclocal for regeneration of Makefiles]),
+ [
+ unset ACLOCAL
+ AM_MISSING_PROG(ACLOCAL, ${withval})
+ AC_MSG_NOTICE([Using $ACLOCAL as aclocal])
+ ])
+])
--- /dev/null
+dnl as-compiler-flag.m4 0.1.0
+
+dnl autostars m4 macro for detection of compiler flags
+
+dnl David Schleef <ds@schleef.org>
+dnl Tim-Philipp Müller <tim centricular net>
+
+dnl AS_COMPILER_FLAG(CFLAGS, ACTION-IF-ACCEPTED, [ACTION-IF-NOT-ACCEPTED])
+dnl Tries to compile with the given CFLAGS.
+dnl Runs ACTION-IF-ACCEPTED if the compiler can compile with the flags,
+dnl and ACTION-IF-NOT-ACCEPTED otherwise.
+
+AC_DEFUN([AS_COMPILER_FLAG],
+[
+ AC_MSG_CHECKING([to see if compiler understands $1])
+
+ save_CFLAGS="$CFLAGS"
+ CFLAGS="$CFLAGS $1"
+
+ AC_TRY_COMPILE([ ], [], [flag_ok=yes], [flag_ok=no])
+ CFLAGS="$save_CFLAGS"
+
+ if test "X$flag_ok" = Xyes ; then
+ $2
+ true
+ else
+ $3
+ true
+ fi
+ AC_MSG_RESULT([$flag_ok])
+])
+
+dnl AS_CXX_COMPILER_FLAG(CPPFLAGS, ACTION-IF-ACCEPTED, [ACTION-IF-NOT-ACCEPTED])
+dnl Tries to compile with the given CPPFLAGS.
+dnl Runs ACTION-IF-ACCEPTED if the compiler can compile with the flags,
+dnl and ACTION-IF-NOT-ACCEPTED otherwise.
+
+AC_DEFUN([AS_CXX_COMPILER_FLAG],
+[
+ AC_REQUIRE([AC_PROG_CXX])
+
+ AC_MSG_CHECKING([to see if c++ compiler understands $1])
+
+ save_CPPFLAGS="$CPPFLAGS"
+ CPPFLAGS="$CPPFLAGS $1"
+
+ AC_LANG_PUSH(C++)
+
+ AC_TRY_COMPILE([ ], [], [flag_ok=yes], [flag_ok=no])
+ CPPFLAGS="$save_CPPFLAGS"
+
+ if test "X$flag_ok" = Xyes ; then
+ $2
+ true
+ else
+ $3
+ true
+ fi
+
+ AC_LANG_POP(C++)
+
+ AC_MSG_RESULT([$flag_ok])
+])
+
+dnl AS_OBJC_COMPILER_FLAG(CPPFLAGS, ACTION-IF-ACCEPTED, [ACTION-IF-NOT-ACCEPTED])
+dnl Tries to compile with the given CPPFLAGS.
+dnl Runs ACTION-IF-ACCEPTED if the compiler can compile with the flags,
+dnl and ACTION-IF-NOT-ACCEPTED otherwise.
+
+AC_DEFUN([AS_OBJC_COMPILER_FLAG],
+[
+ AC_REQUIRE([AC_PROG_OBJC])
+
+ AC_MSG_CHECKING([to see if Objective C compiler understands $1])
+
+ save_CPPFLAGS="$CPPFLAGS"
+ CPPFLAGS="$CPPFLAGS $1"
+
+ AC_LANG_PUSH([Objective C])
+
+ AC_TRY_COMPILE([ ], [], [flag_ok=yes], [flag_ok=no])
+ CPPFLAGS="$save_CPPFLAGS"
+
+ if test "X$flag_ok" = Xyes ; then
+ $2
+ true
+ else
+ $3
+ true
+ fi
+
+ AC_LANG_POP([Objective C])
+
+ AC_MSG_RESULT([$flag_ok])
+])
+
--- /dev/null
+dnl as-compiler.m4 0.1.0
+
+dnl autostars m4 macro for detection of compiler flavor
+
+dnl Thomas Vander Stichele <thomas at apestaart dot org>
+
+dnl $Id: as-compiler.m4,v 1.4 2004/06/01 09:44:19 thomasvs Exp $
+
+dnl AS_COMPILER(COMPILER)
+dnl will set variable COMPILER to
+dnl - gcc
+dnl - forte
+dnl - (empty) if no guess could be made
+
+AC_DEFUN([AS_COMPILER],
+[
+ as_compiler=
+ AC_MSG_CHECKING(for compiler flavour)
+
+ dnl is it gcc ?
+ if test "x$GCC" = "xyes"; then
+ as_compiler="gcc"
+ fi
+
+ dnl is it forte ?
+ AC_TRY_RUN([
+int main
+(int argc, char *argv[])
+{
+#ifdef __sun
+ return 0;
+#else
+ return 1;
+#endif
+}
+ ], as_compiler="forte", ,)
+
+ if test "x$as_compiler" = "x"; then
+ AC_MSG_RESULT([unknown !])
+ else
+ AC_MSG_RESULT($as_compiler)
+ fi
+ [$1]=$as_compiler
+])
--- /dev/null
+dnl AS_DOCBOOK([, ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]])
+dnl checks if xsltproc can build docbook documentation
+dnl (which is possible if the catalog is set up properly
+dnl I also tried checking for a specific version and type of docbook
+dnl but xsltproc seemed to happily run anyway, so we can't check for that
+dnl and version
+dnl this macro takes inspiration from
+dnl http://www.movement.uklinux.net/docs/docbook-autotools/configure.html
+AC_DEFUN([AS_DOCBOOK],
+[
+ XSLTPROC_FLAGS=--nonet
+ DOCBOOK_ROOT=
+ TYPE_LC=xml
+ TYPE_UC=XML
+ DOCBOOK_VERSION=4.1.2
+
+ if test -n "$XML_CATALOG_FILES"; then
+ oldIFS=$IFS
+ IFS=' '
+ for xml_catalog_file in $XML_CATALOG_FILES; do
+ if test -f $xml_catalog_file; then
+ XML_CATALOG=$xml_catalog_file
+ CAT_ENTRY_START='<!--'
+ CAT_ENTRY_END='-->'
+ break
+ fi
+ done
+ IFS=$oldIFS
+ elif test ! -f /etc/xml/catalog; then
+ for i in /usr/share/sgml/docbook/stylesheet/xsl/nwalsh /usr/share/sgml/docbook/xsl-stylesheets/ /usr/local/share/xsl/docbook ;
+ do
+ if test -d "$i"; then
+ DOCBOOK_ROOT=$i
+ fi
+ done
+ else
+ XML_CATALOG=/etc/xml/catalog
+ CAT_ENTRY_START='<!--'
+ CAT_ENTRY_END='-->'
+ fi
+
+ dnl We need xsltproc to process the test
+ AC_CHECK_PROG(XSLTPROC,xsltproc,xsltproc,)
+ XSLTPROC_WORKS=no
+ if test -n "$XSLTPROC"; then
+ AC_MSG_CHECKING([whether xsltproc docbook processing works])
+
+ if test -n "$XML_CATALOG"; then
+ DB_FILE="http://docbook.sourceforge.net/release/xsl/current/xhtml/docbook.xsl"
+ else
+ DB_FILE="$DOCBOOK_ROOT/xhtml/docbook.xsl"
+ fi
+ $XSLTPROC $XSLTPROC_FLAGS $DB_FILE >/dev/null 2>&1 << END
+<?xml version="1.0" encoding='ISO-8859-1'?>
+<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook $TYPE_UC V$DOCBOOK_VERSION//EN" "http://www.oasis-open.org/docbook/$TYPE_LC/$DOCBOOK_VERSION/docbookx.dtd">
+<book id="test">
+</book>
+END
+ if test "$?" = 0; then
+ XSLTPROC_WORKS=yes
+ fi
+ AC_MSG_RESULT($XSLTPROC_WORKS)
+ fi
+
+ if test "x$XSLTPROC_WORKS" = "xyes"; then
+ dnl execute ACTION-IF-FOUND
+ ifelse([$1], , :, [$1])
+ else
+ dnl execute ACTION-IF-NOT-FOUND
+ ifelse([$2], , :, [$2])
+ fi
+
+ AC_SUBST(XML_CATALOG)
+ AC_SUBST(XSLTPROC_FLAGS)
+ AC_SUBST(DOCBOOK_ROOT)
+ AC_SUBST(CAT_ENTRY_START)
+ AC_SUBST(CAT_ENTRY_END)
+])
--- /dev/null
+dnl as-gcc-inline-assembly.m4 0.1.0
+
+dnl autostars m4 macro for detection of gcc inline assembly
+
+dnl David Schleef <ds@schleef.org>
+
+dnl $Id$
+
+dnl AS_COMPILER_FLAG(ACTION-IF-ACCEPTED, [ACTION-IF-NOT-ACCEPTED])
+dnl Tries to compile with the given CFLAGS.
+dnl Runs ACTION-IF-ACCEPTED if the compiler can compile with the flags,
+dnl and ACTION-IF-NOT-ACCEPTED otherwise.
+
+AC_DEFUN([AS_GCC_INLINE_ASSEMBLY],
+[
+ AC_MSG_CHECKING([if compiler supports gcc-style inline assembly])
+
+ AC_TRY_COMPILE([], [
+#ifdef __GNUC_MINOR__
+#if (__GNUC__ * 1000 + __GNUC_MINOR__) < 3004
+#error GCC before 3.4 has critical bugs compiling inline assembly
+#endif
+#endif
+__asm__ (""::) ], [flag_ok=yes], [flag_ok=no])
+
+ if test "X$flag_ok" = Xyes ; then
+ $1
+ true
+ else
+ $2
+ true
+ fi
+ AC_MSG_RESULT([$flag_ok])
+])
+
+
+AC_DEFUN([AS_GCC_ASM_POWERPC_FPU],
+[
+ AC_MSG_CHECKING([if compiler supports FPU instructions on PowerPC])
+
+ AC_TRY_COMPILE([], [__asm__ ("fadd 0,0,0"::) ], [flag_ok=yes], [flag_ok=no])
+
+ if test "X$flag_ok" = Xyes ; then
+ $1
+ true
+ else
+ $2
+ true
+ fi
+ AC_MSG_RESULT([$flag_ok])
+])
+
--- /dev/null
+dnl as-libtool-tags.m4 0.1.4
+
+dnl autostars m4 macro for selecting libtool "tags" (languages)
+
+dnl Andy Wingo does not claim credit for this macro
+dnl backported from libtool 1.6 by Paolo Bonzini
+dnl see http://lists.gnu.org/archive/html/libtool/2003-12/msg00007.html
+
+dnl $Id$
+
+dnl AS_LIBTOOL_TAGS([tags...])
+
+dnl example
+dnl AS_LIBTOOL_TAGS([]) for only C (no fortran, etc)
+
+dnl When AC_LIBTOOL_TAGS is used, I redefine _LT_AC_TAGCONFIG
+dnl to be more similar to the libtool 1.6 implementation, which
+dnl uses an m4 loop and m4 case instead of a shell loop. This
+dnl way the CXX/GCJ/F77/RC tests are not always expanded.
+
+dnl AS_LIBTOOL_TAGS
+dnl ---------------
+dnl tags to enable
+AC_DEFUN([AS_LIBTOOL_TAGS],
+[m4_define([_LT_TAGS],[$1])
+m4_define([_LT_AC_TAGCONFIG], [
+ # redefined LT AC TAGCONFIG
+ if test -f "$ltmain"; then
+ if test ! -f "${ofile}"; then
+ AC_MSG_WARN([output file `$ofile' does not exist])
+ fi
+
+ if test -z "$LTCC"; then
+ eval "`$SHELL ${ofile} --config | grep '^LTCC='`"
+ if test -z "$LTCC"; then
+ AC_MSG_WARN([output file `$ofile' does not look like a libtool script])
+ else
+ AC_MSG_WARN([using `LTCC=$LTCC', extracted from `$ofile'])
+ fi
+ fi
+
+ AC_FOREACH([_LT_TAG], _LT_TAGS,
+ echo THOMAS: tag _LT_TAG
+ [m4_case(_LT_TAG,
+ [CXX], [
+ if test -n "$CXX" && test "X$CXX" != "Xno"; then
+ echo "THOMAS: YAY CXX"
+ AC_LIBTOOL_LANG_CXX_CONFIG
+ available_tags="$available_tags _LT_TAG"
+ fi],
+ [F77], [
+ if test -n "$F77" && test "X$F77" != "Xno"; then
+ AC_LIBTOOL_LANG_F77_CONFIG
+ available_tags="$available_tags _LT_TAG"
+ fi],
+ [GCJ], [
+ if test -n "$GCJ" && test "X$GCJ" != "Xno"; then
+ AC_LIBTOOL_LANG_GCJ_CONFIG
+ available_tags="$available_tags _LT_TAG"
+ fi],
+ [RC], [
+ if test -n "$RC" && test "X$RC" != "Xno"; then
+ AC_LIBTOOL_LANG_RC_CONFIG
+ available_tags="$available_tags _LT_TAG"
+ fi],
+ [m4_errprintn(m4_location[: error: invalid tag name: ]"_LT_TAG")
+ m4_exit(1)])
+ ])
+ echo THOMAS: available tags: $available_tags
+ fi
+ # Now substitute the updated list of available tags.
+ if eval "sed -e 's/^available_tags=.*\$/available_tags=\"$available_tags\"/' \"$ofile\" > \"${ofile}T\""; then
+ mv "${ofile}T" "$ofile"
+ chmod +x "$ofile"
+ AC_MSG_NOTICE([updated available libtool tags with $available_tags.])
+ else
+ rm -f "${ofile}T"
+ AC_MSG_ERROR([unable to update list of available tagged configurations.])
+
+ fi
+
+])dnl _LT_AC_TAG_CONFIG
+])
--- /dev/null
+dnl as-libtool.m4 0.1.4
+
+dnl autostars m4 macro for libtool versioning
+
+dnl Thomas Vander Stichele <thomas at apestaart dot org>
+
+dnl $Id: as-libtool.m4,v 1.10 2005/10/15 13:44:23 thomasvs Exp $
+
+dnl AS_LIBTOOL(PREFIX, CURRENT, REVISION, AGE, [RELEASE])
+
+dnl example
+dnl AS_LIBTOOL(GST, 2, 0, 0)
+
+dnl this macro
+dnl - defines [$PREFIX]_CURRENT, REVISION and AGE
+dnl - defines [$PREFIX]_LIBVERSION
+dnl - defines [$PREFIX]_LT_LDFLAGS to set versioning
+dnl - AC_SUBST's them all
+
+dnl if RELEASE is given, then add a -release option to the LDFLAGS
+dnl with the given release version
+dnl then use [$PREFIX]_LT_LDFLAGS in the relevant Makefile.am's
+
+dnl call AM_PROG_LIBTOOL after this call
+
+AC_DEFUN([AS_LIBTOOL],
+[
+ [$1]_CURRENT=[$2]
+ [$1]_REVISION=[$3]
+ [$1]_AGE=[$4]
+ [$1]_LIBVERSION=[$2]:[$3]:[$4]
+ AC_SUBST([$1]_CURRENT)
+ AC_SUBST([$1]_REVISION)
+ AC_SUBST([$1]_AGE)
+ AC_SUBST([$1]_LIBVERSION)
+
+ [$1]_LT_LDFLAGS="$[$1]_LT_LDFLAGS -version-info $[$1]_LIBVERSION"
+ if test ! -z "[$5]"
+ then
+ [$1]_LT_LDFLAGS="$[$1]_LT_LDFLAGS -release [$5]"
+ fi
+ AC_SUBST([$1]_LT_LDFLAGS)
+
+ LT_PREREQ([2.2.6])
+ LT_INIT([dlopen win32-dll disable-static])
+])
--- /dev/null
+## ------------------------
+## Python file handling
+## From Andrew Dalke
+## Updated by James Henstridge
+## Updated by Andy Wingo to loop through possible pythons
+## ------------------------
+
+# AS_PATH_PYTHON([MINIMUM-VERSION])
+
+# Adds support for distributing Python modules and packages. To
+# install modules, copy them to $(pythondir), using the python_PYTHON
+# automake variable. To install a package with the same name as the
+# automake package, install to $(pkgpythondir), or use the
+# pkgpython_PYTHON automake variable.
+
+# The variables $(pyexecdir) and $(pkgpyexecdir) are provided as
+# locations to install python extension modules (shared libraries).
+# Another macro is required to find the appropriate flags to compile
+# extension modules.
+
+# If your package is configured with a different prefix to python,
+# users will have to add the install directory to the PYTHONPATH
+# environment variable, or create a .pth file (see the python
+# documentation for details).
+
+# If the MINIMUM-VERSION argument is passed, AS_PATH_PYTHON will
+# cause an error if the version of python installed on the system
+# doesn't meet the requirement. MINIMUM-VERSION should consist of
+# numbers and dots only.
+
+# Updated to loop over all possible python binaries by Andy Wingo
+# <wingo@pobox.com>
+# Updated to only warn and unset PYTHON if no good one is found
+
+AC_DEFUN([AS_PATH_PYTHON],
+ [
+ dnl Find a version of Python. I could check for python versions 1.4
+ dnl or earlier, but the default installation locations changed from
+ dnl $prefix/lib/site-python in 1.4 to $prefix/lib/python1.5/site-packages
+ dnl in 1.5, and I don't want to maintain that logic.
+
+ dnl should we do the version check?
+ PYTHON_CANDIDATES="python python2.2 python2.1 python2.0 python2 \
+ python1.6 python1.5"
+ ifelse([$1],[],
+ [AC_PATH_PROG(PYTHON, $PYTHON_CANDIDATES)],
+ [
+ AC_MSG_NOTICE(Looking for Python version >= $1)
+ changequote(<<, >>)dnl
+ prog="
+import sys, string
+minver = '$1'
+# split string by '.' and convert to numeric
+minver_info = map(string.atoi, string.split(minver, '.'))
+# we can now do comparisons on the two lists:
+if sys.version_info >= tuple(minver_info):
+ sys.exit(0)
+else:
+ sys.exit(1)"
+ changequote([, ])dnl
+
+ python_good=false
+ for python_candidate in $PYTHON_CANDIDATES; do
+ unset PYTHON
+ AC_PATH_PROG(PYTHON, $python_candidate) 1> /dev/null 2> /dev/null
+
+ if test "x$PYTHON" = "x"; then continue; fi
+
+ if $PYTHON -c "$prog" 1>&AC_FD_CC 2>&AC_FD_CC; then
+ AC_MSG_CHECKING(["$PYTHON":])
+ AC_MSG_RESULT([okay])
+ python_good=true
+ break;
+ else
+ dnl clear the cache val
+ unset ac_cv_path_PYTHON
+ fi
+ done
+ ])
+
+ if test "$python_good" != "true"; then
+ AC_MSG_WARN([No suitable version of python found])
+ PYTHON=
+ else
+
+ AC_MSG_CHECKING([local Python configuration])
+
+ dnl Query Python for its version number. Getting [:3] seems to be
+ dnl the best way to do this; it's what "site.py" does in the standard
+ dnl library. Need to change quote character because of [:3]
+
+ AC_SUBST(PYTHON_VERSION)
+ changequote(<<, >>)dnl
+ PYTHON_VERSION=`$PYTHON -c "import sys; print sys.version[:3]"`
+ changequote([, ])dnl
+
+
+ dnl Use the values of $prefix and $exec_prefix for the corresponding
+ dnl values of PYTHON_PREFIX and PYTHON_EXEC_PREFIX. These are made
+ dnl distinct variables so they can be overridden if need be. However,
+ dnl general consensus is that you shouldn't need this ability.
+
+ AC_SUBST(PYTHON_PREFIX)
+ PYTHON_PREFIX='${prefix}'
+
+ AC_SUBST(PYTHON_EXEC_PREFIX)
+ PYTHON_EXEC_PREFIX='${exec_prefix}'
+
+ dnl At times (like when building shared libraries) you may want
+ dnl to know which OS platform Python thinks this is.
+
+ AC_SUBST(PYTHON_PLATFORM)
+ PYTHON_PLATFORM=`$PYTHON -c "import sys; print sys.platform"`
+
+
+ dnl Set up 4 directories:
+
+ dnl pythondir -- where to install python scripts. This is the
+ dnl site-packages directory, not the python standard library
+ dnl directory like in previous automake betas. This behaviour
+ dnl is more consistent with lispdir.m4 for example.
+ dnl
+ dnl Also, if the package prefix isn't the same as python's prefix,
+ dnl then the old $(pythondir) was pretty useless.
+
+ AC_SUBST(pythondir)
+ pythondir=$PYTHON_PREFIX"/lib/python"$PYTHON_VERSION/site-packages
+
+ dnl pkgpythondir -- $PACKAGE directory under pythondir. Was
+ dnl PYTHON_SITE_PACKAGE in previous betas, but this naming is
+ dnl more consistent with the rest of automake.
+ dnl Maybe this should be put in python.am?
+
+ AC_SUBST(pkgpythondir)
+ pkgpythondir=\${pythondir}/$PACKAGE
+
+ dnl pyexecdir -- directory for installing python extension modules
+ dnl (shared libraries) Was PYTHON_SITE_EXEC in previous betas.
+
+ AC_SUBST(pyexecdir)
+ pyexecdir=$PYTHON_EXEC_PREFIX"/lib/python"$PYTHON_VERSION/site-packages
+
+ dnl pkgpyexecdir -- $(pyexecdir)/$(PACKAGE)
+ dnl Maybe this should be put in python.am?
+
+ AC_SUBST(pkgpyexecdir)
+ pkgpyexecdir=\${pyexecdir}/$PACKAGE
+
+ AC_MSG_RESULT([looks good])
+
+ fi
+])
--- /dev/null
+dnl as-version.m4 0.2.0
+
+dnl autostars m4 macro for versioning
+
+dnl Thomas Vander Stichele <thomas at apestaart dot org>
+
+dnl $Id: as-version.m4,v 1.15 2006/04/01 09:40:24 thomasvs Exp $
+
+dnl AS_VERSION
+
+dnl example
+dnl AS_VERSION
+
+dnl this macro
+dnl - AC_SUBST's PACKAGE_VERSION_MAJOR, _MINOR, _MICRO
+dnl - AC_SUBST's PACKAGE_VERSION_RELEASE,
+dnl which can be used for rpm release fields
+dnl - doesn't call AM_INIT_AUTOMAKE anymore because it prevents
+dnl maintainer mode from running correctly
+dnl
+dnl don't forget to put #undef PACKAGE_VERSION_RELEASE in acconfig.h
+dnl if you use acconfig.h
+
+AC_DEFUN([AS_VERSION],
+[
+ PACKAGE_VERSION_MAJOR=$(echo AC_PACKAGE_VERSION | cut -d'.' -f1)
+ PACKAGE_VERSION_MINOR=$(echo AC_PACKAGE_VERSION | cut -d'.' -f2)
+ PACKAGE_VERSION_MICRO=$(echo AC_PACKAGE_VERSION | cut -d'.' -f3)
+
+ AC_SUBST(PACKAGE_VERSION_MAJOR)
+ AC_SUBST(PACKAGE_VERSION_MINOR)
+ AC_SUBST(PACKAGE_VERSION_MICRO)
+])
+
+dnl AS_NANO(ACTION-IF-NANO-NON-NULL, [ACTION-IF-NANO-NULL])
+
+dnl requires AC_INIT to be called before
+dnl For projects using a fourth or nano number in your versioning to indicate
+dnl development or prerelease snapshots, this macro allows the build to be
+dnl set up differently accordingly.
+
+dnl this macro:
+dnl - parses AC_PACKAGE_VERSION, set by AC_INIT, and extracts the nano number
+dnl - sets the variable PACKAGE_VERSION_NANO
+dnl - sets the variable PACKAGE_VERSION_RELEASE, which can be used
+dnl for rpm release fields
+dnl - executes ACTION-IF-NANO-NON-NULL or ACTION-IF-NANO-NULL
+
+dnl example:
+dnl AS_NANO(RELEASE="yes", RELEASE="no")
+
+AC_DEFUN([AS_NANO],
+[
+ AC_MSG_CHECKING(nano version)
+
+ NANO=$(echo AC_PACKAGE_VERSION | cut -d'.' -f4)
+
+ if test x"$NANO" = x || test "x$NANO" = "x0" ; then
+ AC_MSG_RESULT([0 (release)])
+ NANO=0
+ PACKAGE_VERSION_RELEASE=1
+ ifelse([$1], , :, [$1])
+ else
+ AC_MSG_RESULT($NANO)
+ PACKAGE_VERSION_RELEASE=0.`date +%Y%m%d.%H%M%S`
+ if test "x$NANO" != "x1" ; then
+ ifelse([$1], , :, [$1])
+ else
+ ifelse([$2], , :, [$2])
+ fi
+ fi
+ PACKAGE_VERSION_NANO=$NANO
+ AC_SUBST(PACKAGE_VERSION_NANO)
+ AC_SUBST(PACKAGE_VERSION_RELEASE)
+])
--- /dev/null
+##### http://autoconf-archive.cryp.to/ax_create_stdint_h.html
+#
+# SYNOPSIS
+#
+# AX_CREATE_STDINT_H [( HEADER-TO-GENERATE [, HEDERS-TO-CHECK])]
+#
+# DESCRIPTION
+#
+# the "ISO C9X: 7.18 Integer types <stdint.h>" section requires the
+# existence of an include file <stdint.h> that defines a set of
+# typedefs, especially uint8_t,int32_t,uintptr_t. Many older
+# installations will not provide this file, but some will have the
+# very same definitions in <inttypes.h>. In other enviroments we can
+# use the inet-types in <sys/types.h> which would define the typedefs
+# int8_t and u_int8_t respectivly.
+#
+# This macros will create a local "_stdint.h" or the headerfile given
+# as an argument. In many cases that file will just "#include
+# <stdint.h>" or "#include <inttypes.h>", while in other environments
+# it will provide the set of basic 'stdint's definitions/typedefs:
+#
+# int8_t,uint8_t,int16_t,uint16_t,int32_t,uint32_t,intptr_t,uintptr_t
+# int_least32_t.. int_fast32_t.. intmax_t
+#
+# which may or may not rely on the definitions of other files, or
+# using the AC_CHECK_SIZEOF macro to determine the actual sizeof each
+# type.
+#
+# if your header files require the stdint-types you will want to
+# create an installable file mylib-int.h that all your other
+# installable header may include. So if you have a library package
+# named "mylib", just use
+#
+# AX_CREATE_STDINT_H(mylib-int.h)
+#
+# in configure.ac and go to install that very header file in
+# Makefile.am along with the other headers (mylib.h) - and the
+# mylib-specific headers can simply use "#include <mylib-int.h>" to
+# obtain the stdint-types.
+#
+# Remember, if the system already had a valid <stdint.h>, the
+# generated file will include it directly. No need for fuzzy
+# HAVE_STDINT_H things... (oops, GCC 4.2.x has deliberatly disabled
+# its stdint.h for non-c99 compilation and the c99-mode is not the
+# default. Therefore this macro will not use the compiler's stdint.h
+# - please complain to the GCC developers).
+#
+# LAST MODIFICATION
+#
+# 2007-06-27
+#
+# COPYLEFT
+#
+# Copyright (c) 2007 Guido U. Draheim <guidod@gmx.de>
+#
+# This program is free software; you can redistribute it and/or
+# modify it under the terms of the GNU General Public License as
+# published by the Free Software Foundation; either version 2 of the
+# License, or (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+# General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
+# 02111-1307, USA.
+#
+# As a special exception, the respective Autoconf Macro's copyright
+# owner gives unlimited permission to copy, distribute and modify the
+# configure scripts that are the output of Autoconf when processing
+# the Macro. You need not follow the terms of the GNU General Public
+# License when using or distributing such scripts, even though
+# portions of the text of the Macro appear in them. The GNU General
+# Public License (GPL) does govern all other use of the material that
+# constitutes the Autoconf Macro.
+#
+# This special exception to the GPL applies to versions of the
+# Autoconf Macro released by the Autoconf Macro Archive. When you
+# make and distribute a modified version of the Autoconf Macro, you
+# may extend this special exception to the GPL to apply to your
+# modified version as well.
+
+AC_DEFUN([AX_CHECK_DATA_MODEL],[
+ AC_CHECK_SIZEOF(char)
+ AC_CHECK_SIZEOF(short)
+ AC_CHECK_SIZEOF(int)
+ AC_CHECK_SIZEOF(long)
+ AC_CHECK_SIZEOF(void*)
+ ac_cv_char_data_model=""
+ ac_cv_char_data_model="$ac_cv_char_data_model$ac_cv_sizeof_char"
+ ac_cv_char_data_model="$ac_cv_char_data_model$ac_cv_sizeof_short"
+ ac_cv_char_data_model="$ac_cv_char_data_model$ac_cv_sizeof_int"
+ ac_cv_long_data_model=""
+ ac_cv_long_data_model="$ac_cv_long_data_model$ac_cv_sizeof_int"
+ ac_cv_long_data_model="$ac_cv_long_data_model$ac_cv_sizeof_long"
+ ac_cv_long_data_model="$ac_cv_long_data_model$ac_cv_sizeof_voidp"
+ AC_MSG_CHECKING([data model])
+ case "$ac_cv_char_data_model/$ac_cv_long_data_model" in
+ 122/242) ac_cv_data_model="IP16" ; n="standard 16bit machine" ;;
+ 122/244) ac_cv_data_model="LP32" ; n="standard 32bit machine" ;;
+ 122/*) ac_cv_data_model="i16" ; n="unusual int16 model" ;;
+ 124/444) ac_cv_data_model="ILP32" ; n="standard 32bit unixish" ;;
+ 124/488) ac_cv_data_model="LP64" ; n="standard 64bit unixish" ;;
+ 124/448) ac_cv_data_model="LLP64" ; n="unusual 64bit unixish" ;;
+ 124/*) ac_cv_data_model="i32" ; n="unusual int32 model" ;;
+ 128/888) ac_cv_data_model="ILP64" ; n="unusual 64bit numeric" ;;
+ 128/*) ac_cv_data_model="i64" ; n="unusual int64 model" ;;
+ 222/*2) ac_cv_data_model="DSP16" ; n="strict 16bit dsptype" ;;
+ 333/*3) ac_cv_data_model="DSP24" ; n="strict 24bit dsptype" ;;
+ 444/*4) ac_cv_data_model="DSP32" ; n="strict 32bit dsptype" ;;
+ 666/*6) ac_cv_data_model="DSP48" ; n="strict 48bit dsptype" ;;
+ 888/*8) ac_cv_data_model="DSP64" ; n="strict 64bit dsptype" ;;
+ 222/*|333/*|444/*|666/*|888/*) :
+ ac_cv_data_model="iDSP" ; n="unusual dsptype" ;;
+ *) ac_cv_data_model="none" ; n="very unusual model" ;;
+ esac
+ AC_MSG_RESULT([$ac_cv_data_model ($ac_cv_long_data_model, $n)])
+])
+
+dnl AX_CHECK_HEADER_STDINT_X([HEADERLIST][,ACTION-IF])
+AC_DEFUN([AX_CHECK_HEADER_STDINT_X],[
+AC_CACHE_CHECK([for stdint uintptr_t], [ac_cv_header_stdint_x],[
+ ac_cv_header_stdint_x="" # the 1997 typedefs (inttypes.h)
+ AC_MSG_RESULT([(..)])
+ for i in m4_ifval([$1],[$1],[stdint.h inttypes.h sys/inttypes.h sys/types.h])
+ do
+ unset ac_cv_type_uintptr_t
+ unset ac_cv_type_uint64_t
+ AC_CHECK_TYPE(uintptr_t,[ac_cv_header_stdint_x=$i],continue,[#include <$i>])
+ AC_CHECK_TYPE(uint64_t,[and64="/uint64_t"],[and64=""],[#include<$i>])
+ m4_ifvaln([$2],[$2]) break
+ done
+ AC_MSG_CHECKING([for stdint uintptr_t])
+ ])
+])
+
+AC_DEFUN([AX_CHECK_HEADER_STDINT_O],[
+AC_CACHE_CHECK([for stdint uint32_t], [ac_cv_header_stdint_o],[
+ ac_cv_header_stdint_o="" # the 1995 typedefs (sys/inttypes.h)
+ AC_MSG_RESULT([(..)])
+ for i in m4_ifval([$1],[$1],[inttypes.h sys/inttypes.h sys/types.h stdint.h])
+ do
+ unset ac_cv_type_uint32_t
+ unset ac_cv_type_uint64_t
+ AC_CHECK_TYPE(uint32_t,[ac_cv_header_stdint_o=$i],continue,[#include <$i>])
+ AC_CHECK_TYPE(uint64_t,[and64="/uint64_t"],[and64=""],[#include<$i>])
+ m4_ifvaln([$2],[$2]) break
+ break;
+ done
+ AC_MSG_CHECKING([for stdint uint32_t])
+ ])
+])
+
+AC_DEFUN([AX_CHECK_HEADER_STDINT_U],[
+AC_CACHE_CHECK([for stdint u_int32_t], [ac_cv_header_stdint_u],[
+ ac_cv_header_stdint_u="" # the BSD typedefs (sys/types.h)
+ AC_MSG_RESULT([(..)])
+ for i in m4_ifval([$1],[$1],[sys/types.h inttypes.h sys/inttypes.h]) ; do
+ unset ac_cv_type_u_int32_t
+ unset ac_cv_type_u_int64_t
+ AC_CHECK_TYPE(u_int32_t,[ac_cv_header_stdint_u=$i],continue,[#include <$i>])
+ AC_CHECK_TYPE(u_int64_t,[and64="/u_int64_t"],[and64=""],[#include<$i>])
+ m4_ifvaln([$2],[$2]) break
+ break;
+ done
+ AC_MSG_CHECKING([for stdint u_int32_t])
+ ])
+])
+
+AC_DEFUN([AX_CREATE_STDINT_H],
+[# ------ AX CREATE STDINT H -------------------------------------
+AC_MSG_CHECKING([for stdint types])
+ac_stdint_h=`echo ifelse($1, , _stdint.h, $1)`
+# try to shortcircuit - if the default include path of the compiler
+# can find a "stdint.h" header then we assume that all compilers can.
+AC_CACHE_VAL([ac_cv_header_stdint_t],[
+old_CXXFLAGS="$CXXFLAGS" ; CXXFLAGS=""
+old_CPPFLAGS="$CPPFLAGS" ; CPPFLAGS=""
+old_CFLAGS="$CFLAGS" ; CFLAGS=""
+AC_TRY_COMPILE([#include <stdint.h>],[int_least32_t v = 0;],
+[ac_cv_stdint_result="(assuming C99 compatible system)"
+ ac_cv_header_stdint_t="stdint.h"; ],
+[ac_cv_header_stdint_t=""])
+if test "$GCC" = "yes" && test ".$ac_cv_header_stdint_t" = "."; then
+CFLAGS="-std=c99"
+AC_TRY_COMPILE([#include <stdint.h>],[int_least32_t v = 0;],
+[AC_MSG_WARN(your GCC compiler has a defunct stdint.h for its default-mode)])
+fi
+CXXFLAGS="$old_CXXFLAGS"
+CPPFLAGS="$old_CPPFLAGS"
+CFLAGS="$old_CFLAGS" ])
+
+v="... $ac_cv_header_stdint_h"
+if test "$ac_stdint_h" = "stdint.h" ; then
+ AC_MSG_RESULT([(are you sure you want them in ./stdint.h?)])
+elif test "$ac_stdint_h" = "inttypes.h" ; then
+ AC_MSG_RESULT([(are you sure you want them in ./inttypes.h?)])
+elif test "_$ac_cv_header_stdint_t" = "_" ; then
+ AC_MSG_RESULT([(putting them into $ac_stdint_h)$v])
+else
+ ac_cv_header_stdint="$ac_cv_header_stdint_t"
+ AC_MSG_RESULT([$ac_cv_header_stdint (shortcircuit)])
+fi
+
+if test "_$ac_cv_header_stdint_t" = "_" ; then # can not shortcircuit..
+
+dnl .....intro message done, now do a few system checks.....
+dnl btw, all old CHECK_TYPE macros do automatically "DEFINE" a type,
+dnl therefore we use the autoconf implementation detail CHECK_TYPE_NEW
+dnl instead that is triggered with 3 or more arguments (see types.m4)
+
+inttype_headers=`echo $2 | sed -e 's/,/ /g'`
+
+ac_cv_stdint_result="(no helpful system typedefs seen)"
+AX_CHECK_HEADER_STDINT_X(dnl
+ stdint.h inttypes.h sys/inttypes.h $inttype_headers,
+ ac_cv_stdint_result="(seen uintptr_t$and64 in $i)")
+
+if test "_$ac_cv_header_stdint_x" = "_" ; then
+AX_CHECK_HEADER_STDINT_O(dnl,
+ inttypes.h sys/inttypes.h stdint.h $inttype_headers,
+ ac_cv_stdint_result="(seen uint32_t$and64 in $i)")
+fi
+
+if test "_$ac_cv_header_stdint_x" = "_" ; then
+if test "_$ac_cv_header_stdint_o" = "_" ; then
+AX_CHECK_HEADER_STDINT_U(dnl,
+ sys/types.h inttypes.h sys/inttypes.h $inttype_headers,
+ ac_cv_stdint_result="(seen u_int32_t$and64 in $i)")
+fi fi
+
+dnl if there was no good C99 header file, do some typedef checks...
+if test "_$ac_cv_header_stdint_x" = "_" ; then
+ AC_MSG_CHECKING([for stdint datatype model])
+ AC_MSG_RESULT([(..)])
+ AX_CHECK_DATA_MODEL
+fi
+
+if test "_$ac_cv_header_stdint_x" != "_" ; then
+ ac_cv_header_stdint="$ac_cv_header_stdint_x"
+elif test "_$ac_cv_header_stdint_o" != "_" ; then
+ ac_cv_header_stdint="$ac_cv_header_stdint_o"
+elif test "_$ac_cv_header_stdint_u" != "_" ; then
+ ac_cv_header_stdint="$ac_cv_header_stdint_u"
+else
+ ac_cv_header_stdint="stddef.h"
+fi
+
+AC_MSG_CHECKING([for extra inttypes in chosen header])
+AC_MSG_RESULT([($ac_cv_header_stdint)])
+dnl see if int_least and int_fast types are present in _this_ header.
+unset ac_cv_type_int_least32_t
+unset ac_cv_type_int_fast32_t
+AC_CHECK_TYPE(int_least32_t,,,[#include <$ac_cv_header_stdint>])
+AC_CHECK_TYPE(int_fast32_t,,,[#include<$ac_cv_header_stdint>])
+AC_CHECK_TYPE(intmax_t,,,[#include <$ac_cv_header_stdint>])
+
+fi # shortcircut to system "stdint.h"
+# ------------------ PREPARE VARIABLES ------------------------------
+if test "$GCC" = "yes" ; then
+ac_cv_stdint_message="using gnu compiler "`$CC --version | head -1`
+else
+ac_cv_stdint_message="using $CC"
+fi
+
+AC_MSG_RESULT([make use of $ac_cv_header_stdint in $ac_stdint_h dnl
+$ac_cv_stdint_result])
+
+dnl -----------------------------------------------------------------
+# ----------------- DONE inttypes.h checks START header -------------
+AC_CONFIG_COMMANDS([$ac_stdint_h],[
+AC_MSG_NOTICE(creating $ac_stdint_h : $_ac_stdint_h)
+ac_stdint=$tmp/_stdint.h
+
+echo "#ifndef" $_ac_stdint_h >$ac_stdint
+echo "#define" $_ac_stdint_h "1" >>$ac_stdint
+echo "#ifndef" _GENERATED_STDINT_H >>$ac_stdint
+echo "#define" _GENERATED_STDINT_H '"'$PACKAGE $VERSION'"' >>$ac_stdint
+echo "/* generated $ac_cv_stdint_message */" >>$ac_stdint
+if test "_$ac_cv_header_stdint_t" != "_" ; then
+echo "#define _STDINT_HAVE_STDINT_H" "1" >>$ac_stdint
+echo "#include <stdint.h>" >>$ac_stdint
+echo "#endif" >>$ac_stdint
+echo "#endif" >>$ac_stdint
+else
+
+cat >>$ac_stdint <<STDINT_EOF
+
+/* ................... shortcircuit part ........................... */
+
+#if defined HAVE_STDINT_H || defined _STDINT_HAVE_STDINT_H
+#include <stdint.h>
+#else
+#include <stddef.h>
+
+/* .................... configured part ............................ */
+
+STDINT_EOF
+
+echo "/* whether we have a C99 compatible stdint header file */" >>$ac_stdint
+if test "_$ac_cv_header_stdint_x" != "_" ; then
+ ac_header="$ac_cv_header_stdint_x"
+ echo "#define _STDINT_HEADER_INTPTR" '"'"$ac_header"'"' >>$ac_stdint
+else
+ echo "/* #undef _STDINT_HEADER_INTPTR */" >>$ac_stdint
+fi
+
+echo "/* whether we have a C96 compatible inttypes header file */" >>$ac_stdint
+if test "_$ac_cv_header_stdint_o" != "_" ; then
+ ac_header="$ac_cv_header_stdint_o"
+ echo "#define _STDINT_HEADER_UINT32" '"'"$ac_header"'"' >>$ac_stdint
+else
+ echo "/* #undef _STDINT_HEADER_UINT32 */" >>$ac_stdint
+fi
+
+echo "/* whether we have a BSD compatible inet types header */" >>$ac_stdint
+if test "_$ac_cv_header_stdint_u" != "_" ; then
+ ac_header="$ac_cv_header_stdint_u"
+ echo "#define _STDINT_HEADER_U_INT32" '"'"$ac_header"'"' >>$ac_stdint
+else
+ echo "/* #undef _STDINT_HEADER_U_INT32 */" >>$ac_stdint
+fi
+
+echo "" >>$ac_stdint
+
+if test "_$ac_header" != "_" ; then if test "$ac_header" != "stddef.h" ; then
+ echo "#include <$ac_header>" >>$ac_stdint
+ echo "" >>$ac_stdint
+fi fi
+
+echo "/* which 64bit typedef has been found */" >>$ac_stdint
+if test "$ac_cv_type_uint64_t" = "yes" ; then
+echo "#define _STDINT_HAVE_UINT64_T" "1" >>$ac_stdint
+else
+echo "/* #undef _STDINT_HAVE_UINT64_T */" >>$ac_stdint
+fi
+if test "$ac_cv_type_u_int64_t" = "yes" ; then
+echo "#define _STDINT_HAVE_U_INT64_T" "1" >>$ac_stdint
+else
+echo "/* #undef _STDINT_HAVE_U_INT64_T */" >>$ac_stdint
+fi
+echo "" >>$ac_stdint
+
+echo "/* which type model has been detected */" >>$ac_stdint
+if test "_$ac_cv_char_data_model" != "_" ; then
+echo "#define _STDINT_CHAR_MODEL" "$ac_cv_char_data_model" >>$ac_stdint
+echo "#define _STDINT_LONG_MODEL" "$ac_cv_long_data_model" >>$ac_stdint
+else
+echo "/* #undef _STDINT_CHAR_MODEL // skipped */" >>$ac_stdint
+echo "/* #undef _STDINT_LONG_MODEL // skipped */" >>$ac_stdint
+fi
+echo "" >>$ac_stdint
+
+echo "/* whether int_least types were detected */" >>$ac_stdint
+if test "$ac_cv_type_int_least32_t" = "yes"; then
+echo "#define _STDINT_HAVE_INT_LEAST32_T" "1" >>$ac_stdint
+else
+echo "/* #undef _STDINT_HAVE_INT_LEAST32_T */" >>$ac_stdint
+fi
+echo "/* whether int_fast types were detected */" >>$ac_stdint
+if test "$ac_cv_type_int_fast32_t" = "yes"; then
+echo "#define _STDINT_HAVE_INT_FAST32_T" "1" >>$ac_stdint
+else
+echo "/* #undef _STDINT_HAVE_INT_FAST32_T */" >>$ac_stdint
+fi
+echo "/* whether intmax_t type was detected */" >>$ac_stdint
+if test "$ac_cv_type_intmax_t" = "yes"; then
+echo "#define _STDINT_HAVE_INTMAX_T" "1" >>$ac_stdint
+else
+echo "/* #undef _STDINT_HAVE_INTMAX_T */" >>$ac_stdint
+fi
+echo "" >>$ac_stdint
+
+ cat >>$ac_stdint <<STDINT_EOF
+/* .................... detections part ............................ */
+
+/* whether we need to define bitspecific types from compiler base types */
+#ifndef _STDINT_HEADER_INTPTR
+#ifndef _STDINT_HEADER_UINT32
+#ifndef _STDINT_HEADER_U_INT32
+#define _STDINT_NEED_INT_MODEL_T
+#else
+#define _STDINT_HAVE_U_INT_TYPES
+#endif
+#endif
+#endif
+
+#ifdef _STDINT_HAVE_U_INT_TYPES
+#undef _STDINT_NEED_INT_MODEL_T
+#endif
+
+#ifdef _STDINT_CHAR_MODEL
+#if _STDINT_CHAR_MODEL+0 == 122 || _STDINT_CHAR_MODEL+0 == 124
+#ifndef _STDINT_BYTE_MODEL
+#define _STDINT_BYTE_MODEL 12
+#endif
+#endif
+#endif
+
+#ifndef _STDINT_HAVE_INT_LEAST32_T
+#define _STDINT_NEED_INT_LEAST_T
+#endif
+
+#ifndef _STDINT_HAVE_INT_FAST32_T
+#define _STDINT_NEED_INT_FAST_T
+#endif
+
+#ifndef _STDINT_HEADER_INTPTR
+#define _STDINT_NEED_INTPTR_T
+#ifndef _STDINT_HAVE_INTMAX_T
+#define _STDINT_NEED_INTMAX_T
+#endif
+#endif
+
+
+/* .................... definition part ............................ */
+
+/* some system headers have good uint64_t */
+#ifndef _HAVE_UINT64_T
+#if defined _STDINT_HAVE_UINT64_T || defined HAVE_UINT64_T
+#define _HAVE_UINT64_T
+#elif defined _STDINT_HAVE_U_INT64_T || defined HAVE_U_INT64_T
+#define _HAVE_UINT64_T
+typedef u_int64_t uint64_t;
+#endif
+#endif
+
+#ifndef _HAVE_UINT64_T
+/* .. here are some common heuristics using compiler runtime specifics */
+#if defined __STDC_VERSION__ && defined __STDC_VERSION__ >= 199901L
+#define _HAVE_UINT64_T
+#define _HAVE_LONGLONG_UINT64_T
+typedef long long int64_t;
+typedef unsigned long long uint64_t;
+
+#elif !defined __STRICT_ANSI__
+#if defined _MSC_VER || defined __WATCOMC__ || defined __BORLANDC__
+#define _HAVE_UINT64_T
+typedef __int64 int64_t;
+typedef unsigned __int64 uint64_t;
+
+#elif defined __GNUC__ || defined __MWERKS__ || defined __ELF__
+/* note: all ELF-systems seem to have loff-support which needs 64-bit */
+#if !defined _NO_LONGLONG
+#define _HAVE_UINT64_T
+#define _HAVE_LONGLONG_UINT64_T
+typedef long long int64_t;
+typedef unsigned long long uint64_t;
+#endif
+
+#elif defined __alpha || (defined __mips && defined _ABIN32)
+#if !defined _NO_LONGLONG
+typedef long int64_t;
+typedef unsigned long uint64_t;
+#endif
+ /* compiler/cpu type to define int64_t */
+#endif
+#endif
+#endif
+
+#if defined _STDINT_HAVE_U_INT_TYPES
+/* int8_t int16_t int32_t defined by inet code, redeclare the u_intXX types */
+typedef u_int8_t uint8_t;
+typedef u_int16_t uint16_t;
+typedef u_int32_t uint32_t;
+
+/* glibc compatibility */
+#ifndef __int8_t_defined
+#define __int8_t_defined
+#endif
+#endif
+
+#ifdef _STDINT_NEED_INT_MODEL_T
+/* we must guess all the basic types. Apart from byte-adressable system, */
+/* there a few 32-bit-only dsp-systems that we guard with BYTE_MODEL 8-} */
+/* (btw, those nibble-addressable systems are way off, or so we assume) */
+
+dnl /* have a look at "64bit and data size neutrality" at */
+dnl /* http://unix.org/version2/whatsnew/login_64bit.html */
+dnl /* (the shorthand "ILP" types always have a "P" part) */
+
+#if defined _STDINT_BYTE_MODEL
+#if _STDINT_LONG_MODEL+0 == 242
+/* 2:4:2 = IP16 = a normal 16-bit system */
+typedef unsigned char uint8_t;
+typedef unsigned short uint16_t;
+typedef unsigned long uint32_t;
+#ifndef __int8_t_defined
+#define __int8_t_defined
+typedef char int8_t;
+typedef short int16_t;
+typedef long int32_t;
+#endif
+#elif _STDINT_LONG_MODEL+0 == 244 || _STDINT_LONG_MODEL == 444
+/* 2:4:4 = LP32 = a 32-bit system derived from a 16-bit */
+/* 4:4:4 = ILP32 = a normal 32-bit system */
+typedef unsigned char uint8_t;
+typedef unsigned short uint16_t;
+typedef unsigned int uint32_t;
+#ifndef __int8_t_defined
+#define __int8_t_defined
+typedef char int8_t;
+typedef short int16_t;
+typedef int int32_t;
+#endif
+#elif _STDINT_LONG_MODEL+0 == 484 || _STDINT_LONG_MODEL+0 == 488
+/* 4:8:4 = IP32 = a 32-bit system prepared for 64-bit */
+/* 4:8:8 = LP64 = a normal 64-bit system */
+typedef unsigned char uint8_t;
+typedef unsigned short uint16_t;
+typedef unsigned int uint32_t;
+#ifndef __int8_t_defined
+#define __int8_t_defined
+typedef char int8_t;
+typedef short int16_t;
+typedef int int32_t;
+#endif
+/* this system has a "long" of 64bit */
+#ifndef _HAVE_UINT64_T
+#define _HAVE_UINT64_T
+typedef unsigned long uint64_t;
+typedef long int64_t;
+#endif
+#elif _STDINT_LONG_MODEL+0 == 448
+/* LLP64 a 64-bit system derived from a 32-bit system */
+typedef unsigned char uint8_t;
+typedef unsigned short uint16_t;
+typedef unsigned int uint32_t;
+#ifndef __int8_t_defined
+#define __int8_t_defined
+typedef char int8_t;
+typedef short int16_t;
+typedef int int32_t;
+#endif
+/* assuming the system has a "long long" */
+#ifndef _HAVE_UINT64_T
+#define _HAVE_UINT64_T
+#define _HAVE_LONGLONG_UINT64_T
+typedef unsigned long long uint64_t;
+typedef long long int64_t;
+#endif
+#else
+#define _STDINT_NO_INT32_T
+#endif
+#else
+#define _STDINT_NO_INT8_T
+#define _STDINT_NO_INT32_T
+#endif
+#endif
+
+/*
+ * quote from SunOS-5.8 sys/inttypes.h:
+ * Use at your own risk. As of February 1996, the committee is squarely
+ * behind the fixed sized types; the "least" and "fast" types are still being
+ * discussed. The probability that the "fast" types may be removed before
+ * the standard is finalized is high enough that they are not currently
+ * implemented.
+ */
+
+#if defined _STDINT_NEED_INT_LEAST_T
+typedef int8_t int_least8_t;
+typedef int16_t int_least16_t;
+typedef int32_t int_least32_t;
+#ifdef _HAVE_UINT64_T
+typedef int64_t int_least64_t;
+#endif
+
+typedef uint8_t uint_least8_t;
+typedef uint16_t uint_least16_t;
+typedef uint32_t uint_least32_t;
+#ifdef _HAVE_UINT64_T
+typedef uint64_t uint_least64_t;
+#endif
+ /* least types */
+#endif
+
+#if defined _STDINT_NEED_INT_FAST_T
+typedef int8_t int_fast8_t;
+typedef int int_fast16_t;
+typedef int32_t int_fast32_t;
+#ifdef _HAVE_UINT64_T
+typedef int64_t int_fast64_t;
+#endif
+
+typedef uint8_t uint_fast8_t;
+typedef unsigned uint_fast16_t;
+typedef uint32_t uint_fast32_t;
+#ifdef _HAVE_UINT64_T
+typedef uint64_t uint_fast64_t;
+#endif
+ /* fast types */
+#endif
+
+#ifdef _STDINT_NEED_INTMAX_T
+#ifdef _HAVE_UINT64_T
+typedef int64_t intmax_t;
+typedef uint64_t uintmax_t;
+#else
+typedef long intmax_t;
+typedef unsigned long uintmax_t;
+#endif
+#endif
+
+#ifdef _STDINT_NEED_INTPTR_T
+#ifndef __intptr_t_defined
+#define __intptr_t_defined
+/* we encourage using "long" to store pointer values, never use "int" ! */
+#if _STDINT_LONG_MODEL+0 == 242 || _STDINT_LONG_MODEL+0 == 484
+typedef unsigned int uintptr_t;
+typedef int intptr_t;
+#elif _STDINT_LONG_MODEL+0 == 244 || _STDINT_LONG_MODEL+0 == 444
+typedef unsigned long uintptr_t;
+typedef long intptr_t;
+#elif _STDINT_LONG_MODEL+0 == 448 && defined _HAVE_UINT64_T
+typedef uint64_t uintptr_t;
+typedef int64_t intptr_t;
+#else /* matches typical system types ILP32 and LP64 - but not IP16 or LLP64 */
+typedef unsigned long uintptr_t;
+typedef long intptr_t;
+#endif
+#endif
+#endif
+
+/* The ISO C99 standard specifies that in C++ implementations these
+ should only be defined if explicitly requested. */
+#if !defined __cplusplus || defined __STDC_CONSTANT_MACROS
+#ifndef UINT32_C
+
+/* Signed. */
+# define INT8_C(c) c
+# define INT16_C(c) c
+# define INT32_C(c) c
+# ifdef _HAVE_LONGLONG_UINT64_T
+# define INT64_C(c) c ## L
+# else
+# define INT64_C(c) c ## LL
+# endif
+
+/* Unsigned. */
+# define UINT8_C(c) c ## U
+# define UINT16_C(c) c ## U
+# define UINT32_C(c) c ## U
+# ifdef _HAVE_LONGLONG_UINT64_T
+# define UINT64_C(c) c ## UL
+# else
+# define UINT64_C(c) c ## ULL
+# endif
+
+/* Maximal type. */
+# ifdef _HAVE_LONGLONG_UINT64_T
+# define INTMAX_C(c) c ## L
+# define UINTMAX_C(c) c ## UL
+# else
+# define INTMAX_C(c) c ## LL
+# define UINTMAX_C(c) c ## ULL
+# endif
+
+ /* literalnumbers */
+#endif
+#endif
+
+/* These limits are merily those of a two complement byte-oriented system */
+
+/* Minimum of signed integral types. */
+# define INT8_MIN (-128)
+# define INT16_MIN (-32767-1)
+# define INT32_MIN (-2147483647-1)
+# define INT64_MIN (-__INT64_C(9223372036854775807)-1)
+/* Maximum of signed integral types. */
+# define INT8_MAX (127)
+# define INT16_MAX (32767)
+# define INT32_MAX (2147483647)
+# define INT64_MAX (__INT64_C(9223372036854775807))
+
+/* Maximum of unsigned integral types. */
+# define UINT8_MAX (255)
+# define UINT16_MAX (65535)
+# define UINT32_MAX (4294967295U)
+# define UINT64_MAX (__UINT64_C(18446744073709551615))
+
+/* Minimum of signed integral types having a minimum size. */
+# define INT_LEAST8_MIN INT8_MIN
+# define INT_LEAST16_MIN INT16_MIN
+# define INT_LEAST32_MIN INT32_MIN
+# define INT_LEAST64_MIN INT64_MIN
+/* Maximum of signed integral types having a minimum size. */
+# define INT_LEAST8_MAX INT8_MAX
+# define INT_LEAST16_MAX INT16_MAX
+# define INT_LEAST32_MAX INT32_MAX
+# define INT_LEAST64_MAX INT64_MAX
+
+/* Maximum of unsigned integral types having a minimum size. */
+# define UINT_LEAST8_MAX UINT8_MAX
+# define UINT_LEAST16_MAX UINT16_MAX
+# define UINT_LEAST32_MAX UINT32_MAX
+# define UINT_LEAST64_MAX UINT64_MAX
+
+ /* shortcircuit*/
+#endif
+ /* once */
+#endif
+#endif
+STDINT_EOF
+fi
+ if cmp -s $ac_stdint_h $ac_stdint 2>/dev/null; then
+ AC_MSG_NOTICE([$ac_stdint_h is unchanged])
+ else
+ ac_dir=`AS_DIRNAME(["$ac_stdint_h"])`
+ AS_MKDIR_P(["$ac_dir"])
+ rm -f $ac_stdint_h
+ mv $ac_stdint $ac_stdint_h
+ fi
+],[# variables for create stdint.h replacement
+PACKAGE="$PACKAGE"
+VERSION="$VERSION"
+ac_stdint_h="$ac_stdint_h"
+_ac_stdint_h=AS_TR_CPP(_$PACKAGE-$ac_stdint_h)
+ac_cv_stdint_message="$ac_cv_stdint_message"
+ac_cv_header_stdint_t="$ac_cv_header_stdint_t"
+ac_cv_header_stdint_x="$ac_cv_header_stdint_x"
+ac_cv_header_stdint_o="$ac_cv_header_stdint_o"
+ac_cv_header_stdint_u="$ac_cv_header_stdint_u"
+ac_cv_type_uint64_t="$ac_cv_type_uint64_t"
+ac_cv_type_u_int64_t="$ac_cv_type_u_int64_t"
+ac_cv_char_data_model="$ac_cv_char_data_model"
+ac_cv_long_data_model="$ac_cv_long_data_model"
+ac_cv_type_int_least32_t="$ac_cv_type_int_least32_t"
+ac_cv_type_int_fast32_t="$ac_cv_type_int_fast32_t"
+ac_cv_type_intmax_t="$ac_cv_type_intmax_t"
+])
+])
--- /dev/null
+dnl _AM_TRY_CHECK(MINIMUM-VERSION, EXTRA-CFLAGS, EXTRA-LIBS, CHECK-LIB-NAME
+dnl [, ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]])
+dnl Test for check, and define CHECK_CFLAGS and CHECK_LIBS
+dnl Done this way because of the brokenness that is
+dnl https://launchpad.net/distros/ubuntu/+source/check/+bug/5840
+dnl
+
+AC_DEFUN([_AM_TRY_CHECK],
+[
+ min_check_version=$1
+ extra_cflags=$2
+ extra_libs=$3
+ check_lib_name=$4
+
+ CHECK_CFLAGS="$extra_cflags"
+ CHECK_LIBS="$extra_libs -l$check_lib_name"
+
+ ac_save_CFLAGS="$CFLAGS"
+ ac_save_LIBS="$LIBS"
+
+ CFLAGS="$CFLAGS $CHECK_CFLAGS"
+ LIBS="$CHECK_LIBS $LIBS"
+
+ AC_MSG_CHECKING(for check named $check_lib_name - version >= $min_check_version)
+
+ rm -f conf.check-test
+ dnl unset no_check, since in our second run it would have been set to yes
+ dnl before
+ no_check=
+ AC_TRY_RUN([
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <check.h>
+
+int main ()
+{
+ int major, minor, micro;
+ char *tmp_version;
+
+ system ("touch conf.check-test");
+
+ /* HP/UX 9 (%@#!) writes to sscanf strings */
+ tmp_version = strdup("$min_check_version");
+ if (sscanf(tmp_version, "%d.%d.%d", &major, &minor, µ) != 3) {
+ printf("%s, bad version string\n", "$min_check_version");
+ return 1;
+ }
+
+ if ((CHECK_MAJOR_VERSION != check_major_version) ||
+ (CHECK_MINOR_VERSION != check_minor_version) ||
+ (CHECK_MICRO_VERSION != check_micro_version))
+ {
+ printf("\n*** The check header file (version %d.%d.%d) does not match\n",
+ CHECK_MAJOR_VERSION, CHECK_MINOR_VERSION, CHECK_MICRO_VERSION);
+ printf("*** the check library (version %d.%d.%d).\n",
+ check_major_version, check_minor_version, check_micro_version);
+ return 1;
+ }
+
+ if ((check_major_version > major) ||
+ ((check_major_version == major) && (check_minor_version > minor)) ||
+ ((check_major_version == major) && (check_minor_version == minor) && (check_micro_version >= micro)))
+ {
+ return 0;
+ }
+ else
+ {
+ printf("\n*** An old version of check (%d.%d.%d) was found.\n",
+ check_major_version, check_minor_version, check_micro_version);
+ printf("*** You need a version of check being at least %d.%d.%d.\n", major, minor, micro);
+ printf("***\n");
+ printf("*** If you have already installed a sufficiently new version, this error\n");
+ printf("*** probably means that the wrong copy of the check library and header\n");
+ printf("*** file is being found. Rerun configure with the --with-check=PATH option\n");
+ printf("*** to specify the prefix where the correct version was installed.\n");
+ }
+
+ return 1;
+}
+],, no_check=yes, [echo $ac_n "cross compiling; assumed OK... $ac_c"])
+
+ CFLAGS="$ac_save_CFLAGS"
+ LIBS="$ac_save_LIBS"
+
+ if test "x$no_check" = x ; then
+ AC_MSG_RESULT(yes)
+ ifelse([$5], , :, [$5])
+ else
+ AC_MSG_RESULT(no)
+ if test -f conf.check-test ; then
+ :
+ else
+ echo "*** Could not run check test program, checking why..."
+ CFLAGS="$CFLAGS $CHECK_CFLAGS"
+ LIBS="$CHECK_LIBS $LIBS"
+ AC_TRY_LINK([
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <check.h>
+], , [ echo "*** The test program compiled, but did not run. This usually means"
+ echo "*** that the run-time linker is not finding check. You'll need to set your"
+ echo "*** LD_LIBRARY_PATH environment variable, or edit /etc/ld.so.conf to point"
+ echo "*** to the installed location Also, make sure you have run ldconfig if that"
+ echo "*** is required on your system"
+ echo "***"
+ echo "*** If you have an old version installed, it is best to remove it, although"
+ echo "*** you may also be able to get things to work by modifying LD_LIBRARY_PATH"],
+ [ echo "*** The test program failed to compile or link. See the file config.log for"
+ echo "*** the exact error that occured." ])
+
+ CFLAGS="$ac_save_CFLAGS"
+ LIBS="$ac_save_LIBS"
+ fi
+
+ CHECK_CFLAGS=""
+ CHECK_LIBS=""
+
+ rm -f conf.check-test
+ ifelse([$6], , AC_MSG_ERROR([check not found]), [$6])
+ fi
+])
+
+
+dnl AM_PATH_CHECK([MINIMUM-VERSION, [ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]]])
+dnl Test for check, and define CHECK_CFLAGS and CHECK_LIBS
+dnl
+
+AC_DEFUN([AM_PATH_CHECK],
+[
+ AC_ARG_WITH(check,
+ [ --with-check=PATH prefix where check is installed [default=auto]])
+
+ AC_ARG_WITH(checklibname,
+ AC_HELP_STRING([--with-check-lib-name=NAME],
+ [name of the PIC check library (default=check)]))
+
+ min_check_version=ifelse([$1], ,0.8.2,$1)
+
+ if test x$with_check = xno; then
+ AC_MSG_RESULT(disabled)
+ ifelse([$3], , AC_MSG_ERROR([disabling check is not supported]), [$3])
+ else
+ if test "x$with_check" != x; then
+ CHECK_EXTRA_CFLAGS="-I$with_check/include"
+ CHECK_EXTRA_LIBS="-L$with_check/lib"
+ else
+ CHECK_EXTRA_CFLAGS=""
+ CHECK_EXTRA_LIBS=""
+ fi
+
+ if test x$with_checklibname = x; then
+ _AM_TRY_CHECK($min_check_version, $CHECK_EXTRA_CFLAGS, $CHECK_EXTRA_LIBS,
+ check_pic, [have_check=true], [have_check=false])
+ if test x$have_check = xtrue; then
+ ifelse([$2], , :, [$2])
+ else
+ _AM_TRY_CHECK($min_check_version, $CHECK_EXTRA_CFLAGS, $CHECK_EXTRA_LIBS,
+ check, [have_check=true], [have_check=false])
+ if test x$have_check = xtrue; then
+ ifelse([$2], , :, [$2])
+ else
+ ifelse([$3], , AC_MSG_ERROR([check not found]), [$3])
+ fi
+ fi
+ else
+ _AM_TRY_CHECK($min_check_version, $CHECK_EXTRA_CFLAGS, $CHECK_EXTRA_LIBS,
+ $with_checklibname, [have_check=true], [have_check=false])
+ if test x$have_check = xtrue; then
+ ifelse([$2], , :, [$2])
+ else
+ ifelse([$3], , AC_MSG_ERROR([check not found]), [$3])
+ fi
+ fi
+
+ AC_SUBST(CHECK_CFLAGS)
+ AC_SUBST(CHECK_LIBS)
+ rm -f conf.check-test
+ fi
+])
--- /dev/null
+# Copyright (C) 1995-2002 Free Software Foundation, Inc.
+# Copyright (C) 2001-2003,2004 Red Hat, Inc.
+#
+# This file is free software, distributed under the terms of the GNU
+# General Public License. As a special exception to the GNU General
+# Public License, this file may be distributed as part of a program
+# that contains a configuration script generated by Autoconf, under
+# the same distribution terms as the rest of that program.
+#
+# This file can be copied and used freely without restrictions. It can
+# be used in projects which are not available under the GNU Public License
+# but which still want to provide support for the GNU gettext functionality.
+#
+# Macro to add for using GNU gettext.
+# Ulrich Drepper <drepper@cygnus.com>, 1995, 1996
+#
+# Modified to never use included libintl.
+# Owen Taylor <otaylor@redhat.com>, 12/15/1998
+#
+# Major rework to remove unused code
+# Owen Taylor <otaylor@redhat.com>, 12/11/2002
+#
+# Added better handling of ALL_LINGUAS from GNU gettext version
+# written by Bruno Haible, Owen Taylor <otaylor.redhat.com> 5/30/3002
+#
+# Modified to require ngettext
+# Matthias Clasen <mclasen@redhat.com> 08/06/2004
+#
+# We need this here as well, since someone might use autoconf-2.5x
+# to configure GLib then an older version to configure a package
+# using AM_GLIB_GNU_GETTEXT
+AC_PREREQ(2.53)
+
+dnl
+dnl We go to great lengths to make sure that aclocal won't
+dnl try to pull in the installed version of these macros
+dnl when running aclocal in the glib directory.
+dnl
+m4_copy([AC_DEFUN],[glib_DEFUN])
+m4_copy([AC_REQUIRE],[glib_REQUIRE])
+dnl
+dnl At the end, if we're not within glib, we'll define the public
+dnl definitions in terms of our private definitions.
+dnl
+
+# GLIB_LC_MESSAGES
+#--------------------
+glib_DEFUN([GLIB_LC_MESSAGES],
+ [AC_CHECK_HEADERS([locale.h])
+ if test $ac_cv_header_locale_h = yes; then
+ AC_CACHE_CHECK([for LC_MESSAGES], am_cv_val_LC_MESSAGES,
+ [AC_TRY_LINK([#include <locale.h>], [return LC_MESSAGES],
+ am_cv_val_LC_MESSAGES=yes, am_cv_val_LC_MESSAGES=no)])
+ if test $am_cv_val_LC_MESSAGES = yes; then
+ AC_DEFINE(HAVE_LC_MESSAGES, 1,
+ [Define if your <locale.h> file defines LC_MESSAGES.])
+ fi
+ fi])
+
+# GLIB_PATH_PROG_WITH_TEST
+#----------------------------
+dnl GLIB_PATH_PROG_WITH_TEST(VARIABLE, PROG-TO-CHECK-FOR,
+dnl TEST-PERFORMED-ON-FOUND_PROGRAM [, VALUE-IF-NOT-FOUND [, PATH]])
+glib_DEFUN([GLIB_PATH_PROG_WITH_TEST],
+[# Extract the first word of "$2", so it can be a program name with args.
+set dummy $2; ac_word=[$]2
+AC_MSG_CHECKING([for $ac_word])
+AC_CACHE_VAL(ac_cv_path_$1,
+[case "[$]$1" in
+ /*)
+ ac_cv_path_$1="[$]$1" # Let the user override the test with a path.
+ ;;
+ *)
+ IFS="${IFS= }"; ac_save_ifs="$IFS"; IFS="${IFS}:"
+ for ac_dir in ifelse([$5], , $PATH, [$5]); do
+ test -z "$ac_dir" && ac_dir=.
+ if test -f $ac_dir/$ac_word; then
+ if [$3]; then
+ ac_cv_path_$1="$ac_dir/$ac_word"
+ break
+ fi
+ fi
+ done
+ IFS="$ac_save_ifs"
+dnl If no 4th arg is given, leave the cache variable unset,
+dnl so AC_PATH_PROGS will keep looking.
+ifelse([$4], , , [ test -z "[$]ac_cv_path_$1" && ac_cv_path_$1="$4"
+])dnl
+ ;;
+esac])dnl
+$1="$ac_cv_path_$1"
+if test ifelse([$4], , [-n "[$]$1"], ["[$]$1" != "$4"]); then
+ AC_MSG_RESULT([$]$1)
+else
+ AC_MSG_RESULT(no)
+fi
+AC_SUBST($1)dnl
+])
+
+# GLIB_WITH_NLS
+#-----------------
+glib_DEFUN([GLIB_WITH_NLS],
+ dnl NLS is obligatory
+ [USE_NLS=yes
+ AC_SUBST(USE_NLS)
+
+ gt_cv_have_gettext=no
+
+ CATOBJEXT=NONE
+ XGETTEXT=:
+ INTLLIBS=
+
+ AC_CHECK_HEADER(libintl.h,
+ [gt_cv_func_dgettext_libintl="no"
+ libintl_extra_libs=""
+
+ #
+ # First check in libc
+ #
+ AC_CACHE_CHECK([for ngettext in libc], gt_cv_func_ngettext_libc,
+ [AC_TRY_LINK([
+#include <libintl.h>
+],
+ [return !ngettext ("","", 1)],
+ gt_cv_func_ngettext_libc=yes,
+ gt_cv_func_ngettext_libc=no)
+ ])
+
+ if test "$gt_cv_func_ngettext_libc" = "yes" ; then
+ AC_CACHE_CHECK([for dgettext in libc], gt_cv_func_dgettext_libc,
+ [AC_TRY_LINK([
+#include <libintl.h>
+],
+ [return !dgettext ("","")],
+ gt_cv_func_dgettext_libc=yes,
+ gt_cv_func_dgettext_libc=no)
+ ])
+ fi
+
+ if test "$gt_cv_func_ngettext_libc" = "yes" ; then
+ AC_CHECK_FUNCS(bind_textdomain_codeset)
+ fi
+
+ #
+ # If we don't have everything we want, check in libintl
+ #
+ if test "$gt_cv_func_dgettext_libc" != "yes" \
+ || test "$gt_cv_func_ngettext_libc" != "yes" \
+ || test "$ac_cv_func_bind_textdomain_codeset" != "yes" ; then
+
+ AC_CHECK_LIB(intl, bindtextdomain,
+ [AC_CHECK_LIB(intl, ngettext,
+ [AC_CHECK_LIB(intl, dgettext,
+ gt_cv_func_dgettext_libintl=yes)])])
+
+ if test "$gt_cv_func_dgettext_libintl" != "yes" ; then
+ AC_MSG_CHECKING([if -liconv is needed to use gettext])
+ AC_MSG_RESULT([])
+ AC_CHECK_LIB(intl, ngettext,
+ [AC_CHECK_LIB(intl, dcgettext,
+ [gt_cv_func_dgettext_libintl=yes
+ libintl_extra_libs=-liconv],
+ :,-liconv)],
+ :,-liconv)
+ fi
+
+ #
+ # If we found libintl, then check in it for bind_textdomain_codeset();
+ # we'll prefer libc if neither have bind_textdomain_codeset(),
+ # and both have dgettext and ngettext
+ #
+ if test "$gt_cv_func_dgettext_libintl" = "yes" ; then
+ glib_save_LIBS="$LIBS"
+ LIBS="$LIBS -lintl $libintl_extra_libs"
+ unset ac_cv_func_bind_textdomain_codeset
+ AC_CHECK_FUNCS(bind_textdomain_codeset)
+ LIBS="$glib_save_LIBS"
+
+ if test "$ac_cv_func_bind_textdomain_codeset" = "yes" ; then
+ gt_cv_func_dgettext_libc=no
+ else
+ if test "$gt_cv_func_dgettext_libc" = "yes" \
+ && test "$gt_cv_func_ngettext_libc" = "yes"; then
+ gt_cv_func_dgettext_libintl=no
+ fi
+ fi
+ fi
+ fi
+
+ if test "$gt_cv_func_dgettext_libc" = "yes" \
+ || test "$gt_cv_func_dgettext_libintl" = "yes"; then
+ gt_cv_have_gettext=yes
+ fi
+
+ if test "$gt_cv_func_dgettext_libintl" = "yes"; then
+ INTLLIBS="-lintl $libintl_extra_libs"
+ fi
+
+ if test "$gt_cv_have_gettext" = "yes"; then
+ AC_DEFINE(HAVE_GETTEXT,1,
+ [Define if the GNU gettext() function is already present or preinstalled.])
+ GLIB_PATH_PROG_WITH_TEST(MSGFMT, msgfmt,
+ [test -z "`$ac_dir/$ac_word -h 2>&1 | grep 'dv '`"], no)dnl
+ if test "$MSGFMT" != "no"; then
+ glib_save_LIBS="$LIBS"
+ LIBS="$LIBS $INTLLIBS"
+ AC_CHECK_FUNCS(dcgettext)
+ MSGFMT_OPTS=
+ AC_MSG_CHECKING([if msgfmt accepts -c])
+ GLIB_RUN_PROG([$MSGFMT -c -o /dev/null],[
+msgid ""
+msgstr ""
+"Content-Type: text/plain; charset=UTF-8\n"
+"Project-Id-Version: test 1.0\n"
+"PO-Revision-Date: 2007-02-15 12:01+0100\n"
+"Last-Translator: test <foo@bar.xx>\n"
+"Language-Team: C <LL@li.org>\n"
+"MIME-Version: 1.0\n"
+"Content-Transfer-Encoding: 8bit\n"
+], [MSGFMT_OPTS=-c; AC_MSG_RESULT([yes])], [AC_MSG_RESULT([no])])
+ AC_SUBST(MSGFMT_OPTS)
+ AC_PATH_PROG(GMSGFMT, gmsgfmt, $MSGFMT)
+ GLIB_PATH_PROG_WITH_TEST(XGETTEXT, xgettext,
+ [test -z "`$ac_dir/$ac_word -h 2>&1 | grep '(HELP)'`"], :)
+ AC_TRY_LINK(, [extern int _nl_msg_cat_cntr;
+ return _nl_msg_cat_cntr],
+ [CATOBJEXT=.gmo
+ DATADIRNAME=share],
+ [case $host in
+ *-*-solaris*)
+ dnl On Solaris, if bind_textdomain_codeset is in libc,
+ dnl GNU format message catalog is always supported,
+ dnl since both are added to the libc all together.
+ dnl Hence, we'd like to go with DATADIRNAME=share and
+ dnl and CATOBJEXT=.gmo in this case.
+ AC_CHECK_FUNC(bind_textdomain_codeset,
+ [CATOBJEXT=.gmo
+ DATADIRNAME=share],
+ [CATOBJEXT=.mo
+ DATADIRNAME=lib])
+ ;;
+ *)
+ CATOBJEXT=.mo
+ DATADIRNAME=lib
+ ;;
+ esac])
+ LIBS="$glib_save_LIBS"
+ INSTOBJEXT=.mo
+ else
+ gt_cv_have_gettext=no
+ fi
+ fi
+ ])
+
+ if test "$gt_cv_have_gettext" = "yes" ; then
+ AC_DEFINE(ENABLE_NLS, 1,
+ [always defined to indicate that i18n is enabled])
+ fi
+
+ dnl Test whether we really found GNU xgettext.
+ if test "$XGETTEXT" != ":"; then
+ dnl If it is not GNU xgettext we define it as : so that the
+ dnl Makefiles still can work.
+ if $XGETTEXT --omit-header /dev/null 2> /dev/null; then
+ : ;
+ else
+ AC_MSG_RESULT(
+ [found xgettext program is not GNU xgettext; ignore it])
+ XGETTEXT=":"
+ fi
+ fi
+
+ # We need to process the po/ directory.
+ POSUB=po
+
+ AC_OUTPUT_COMMANDS(
+ [case "$CONFIG_FILES" in *po/Makefile.in*)
+ sed -e "/POTFILES =/r po/POTFILES" po/Makefile.in > po/Makefile
+ esac])
+
+ dnl These rules are solely for the distribution goal. While doing this
+ dnl we only have to keep exactly one list of the available catalogs
+ dnl in configure.in.
+ for lang in $ALL_LINGUAS; do
+ GMOFILES="$GMOFILES $lang.gmo"
+ POFILES="$POFILES $lang.po"
+ done
+
+ dnl Make all variables we use known to autoconf.
+ AC_SUBST(CATALOGS)
+ AC_SUBST(CATOBJEXT)
+ AC_SUBST(DATADIRNAME)
+ AC_SUBST(GMOFILES)
+ AC_SUBST(INSTOBJEXT)
+ AC_SUBST(INTLLIBS)
+ AC_SUBST(PO_IN_DATADIR_TRUE)
+ AC_SUBST(PO_IN_DATADIR_FALSE)
+ AC_SUBST(POFILES)
+ AC_SUBST(POSUB)
+ ])
+
+# AM_GLIB_GNU_GETTEXT
+# -------------------
+# Do checks necessary for use of gettext. If a suitable implementation
+# of gettext is found in either in libintl or in the C library,
+# it will set INTLLIBS to the libraries needed for use of gettext
+# and AC_DEFINE() HAVE_GETTEXT and ENABLE_NLS. (The shell variable
+# gt_cv_have_gettext will be set to "yes".) It will also call AC_SUBST()
+# on various variables needed by the Makefile.in.in installed by
+# glib-gettextize.
+dnl
+glib_DEFUN([GLIB_GNU_GETTEXT],
+ [AC_REQUIRE([AC_PROG_CC])dnl
+ AC_REQUIRE([AC_HEADER_STDC])dnl
+
+ GLIB_LC_MESSAGES
+ GLIB_WITH_NLS
+
+ if test "$gt_cv_have_gettext" = "yes"; then
+ if test "x$ALL_LINGUAS" = "x"; then
+ LINGUAS=
+ else
+ AC_MSG_CHECKING(for catalogs to be installed)
+ NEW_LINGUAS=
+ for presentlang in $ALL_LINGUAS; do
+ useit=no
+ if test "%UNSET%" != "${LINGUAS-%UNSET%}"; then
+ desiredlanguages="$LINGUAS"
+ else
+ desiredlanguages="$ALL_LINGUAS"
+ fi
+ for desiredlang in $desiredlanguages; do
+ # Use the presentlang catalog if desiredlang is
+ # a. equal to presentlang, or
+ # b. a variant of presentlang (because in this case,
+ # presentlang can be used as a fallback for messages
+ # which are not translated in the desiredlang catalog).
+ case "$desiredlang" in
+ "$presentlang"*) useit=yes;;
+ esac
+ done
+ if test $useit = yes; then
+ NEW_LINGUAS="$NEW_LINGUAS $presentlang"
+ fi
+ done
+ LINGUAS=$NEW_LINGUAS
+ AC_MSG_RESULT($LINGUAS)
+ fi
+
+ dnl Construct list of names of catalog files to be constructed.
+ if test -n "$LINGUAS"; then
+ for lang in $LINGUAS; do CATALOGS="$CATALOGS $lang$CATOBJEXT"; done
+ fi
+ fi
+
+ dnl If the AC_CONFIG_AUX_DIR macro for autoconf is used we possibly
+ dnl find the mkinstalldirs script in another subdir but ($top_srcdir).
+ dnl Try to locate is.
+ MKINSTALLDIRS=
+ if test -n "$ac_aux_dir"; then
+ MKINSTALLDIRS="$ac_aux_dir/mkinstalldirs"
+ fi
+ if test -z "$MKINSTALLDIRS"; then
+ MKINSTALLDIRS="\$(top_srcdir)/mkinstalldirs"
+ fi
+ AC_SUBST(MKINSTALLDIRS)
+
+ dnl Generate list of files to be processed by xgettext which will
+ dnl be included in po/Makefile.
+ test -d po || mkdir po
+ if test "x$srcdir" != "x."; then
+ if test "x`echo $srcdir | sed 's@/.*@@'`" = "x"; then
+ posrcprefix="$srcdir/"
+ else
+ posrcprefix="../$srcdir/"
+ fi
+ else
+ posrcprefix="../"
+ fi
+ rm -f po/POTFILES
+ sed -e "/^#/d" -e "/^\$/d" -e "s,.*, $posrcprefix& \\\\," -e "\$s/\(.*\) \\\\/\1/" \
+ < $srcdir/po/POTFILES.in > po/POTFILES
+ ])
+
+# AM_GLIB_DEFINE_LOCALEDIR(VARIABLE)
+# -------------------------------
+# Define VARIABLE to the location where catalog files will
+# be installed by po/Makefile.
+glib_DEFUN([GLIB_DEFINE_LOCALEDIR],
+[glib_REQUIRE([GLIB_GNU_GETTEXT])dnl
+glib_save_prefix="$prefix"
+glib_save_exec_prefix="$exec_prefix"
+glib_save_datarootdir="$datarootdir"
+test "x$prefix" = xNONE && prefix=$ac_default_prefix
+test "x$exec_prefix" = xNONE && exec_prefix=$prefix
+datarootdir=`eval echo "${datarootdir}"`
+if test "x$CATOBJEXT" = "x.mo" ; then
+ localedir=`eval echo "${libdir}/locale"`
+else
+ localedir=`eval echo "${datadir}/locale"`
+fi
+prefix="$glib_save_prefix"
+exec_prefix="$glib_save_exec_prefix"
+datarootdir="$glib_save_datarootdir"
+AC_DEFINE_UNQUOTED($1, "$localedir",
+ [Define the location where the catalogs will be installed])
+])
+
+dnl
+dnl Now the definitions that aclocal will find
+dnl
+ifdef(glib_configure_in,[],[
+AC_DEFUN([AM_GLIB_GNU_GETTEXT],[GLIB_GNU_GETTEXT($@)])
+AC_DEFUN([AM_GLIB_DEFINE_LOCALEDIR],[GLIB_DEFINE_LOCALEDIR($@)])
+])dnl
+
+# GLIB_RUN_PROG(PROGRAM, TEST-FILE, [ACTION-IF-PASS], [ACTION-IF-FAIL])
+#
+# Create a temporary file with TEST-FILE as its contents and pass the
+# file name to PROGRAM. Perform ACTION-IF-PASS if PROGRAM exits with
+# 0 and perform ACTION-IF-FAIL for any other exit status.
+AC_DEFUN([GLIB_RUN_PROG],
+[cat >conftest.foo <<_ACEOF
+$2
+_ACEOF
+if AC_RUN_LOG([$1 conftest.foo]); then
+ m4_ifval([$3], [$3], [:])
+m4_ifvaln([$4], [else $4])dnl
+echo "$as_me: failed input was:" >&AS_MESSAGE_LOG_FD
+sed 's/^/| /' conftest.foo >&AS_MESSAGE_LOG_FD
+fi])
+
--- /dev/null
+dnl AG_GST_ARCH
+dnl sets up defines and automake conditionals for host architecture
+dnl checks endianness
+dnl defines HOST_CPU
+
+AC_DEFUN([AG_GST_ARCH],
+[
+ dnl Determine CPU
+ case "x${target_cpu}" in
+ xi?86 | xk? | xi?86_64)
+ case $target_os in
+ solaris*)
+ AC_CHECK_DECL([__i386], [I386_ABI="yes"], [I386_ABI="no"])
+ AC_CHECK_DECL([__amd64], [AMD64_ABI="yes"], [AMD64_ABI="no"])
+
+ if test "x$I386_ABI" = "xyes" ; then
+ HAVE_CPU_I386=yes
+ AC_DEFINE(HAVE_CPU_I386, 1, [Define if the target CPU is an x86])
+ fi
+ if test "x$AMD64_ABI" = "xyes" ; then
+ HAVE_CPU_X86_64=yes
+ AC_DEFINE(HAVE_CPU_X86_64, 1, [Define if the target CPU is a x86_64])
+ fi
+ ;;
+ *)
+ HAVE_CPU_I386=yes
+ AC_DEFINE(HAVE_CPU_I386, 1, [Define if the target CPU is an x86])
+
+ dnl FIXME could use some better detection
+ dnl (ie CPUID)
+ case "x${target_cpu}" in
+ xi386 | xi486) ;;
+ *)
+ AC_DEFINE(HAVE_RDTSC, 1, [Define if RDTSC is available]) ;;
+ esac
+ ;;
+ esac
+ ;;
+ xpowerpc)
+ HAVE_CPU_PPC=yes
+ AC_DEFINE(HAVE_CPU_PPC, 1, [Define if the target CPU is a PowerPC]) ;;
+ xpowerpc64)
+ HAVE_CPU_PPC64=yes
+ AC_DEFINE(HAVE_CPU_PPC64, 1, [Define if the target CPU is a 64 bit PowerPC]) ;;
+ xalpha*)
+ HAVE_CPU_ALPHA=yes
+ AC_DEFINE(HAVE_CPU_ALPHA, 1, [Define if the target CPU is an Alpha]) ;;
+ xarm*)
+ HAVE_CPU_ARM=yes
+ AC_DEFINE(HAVE_CPU_ARM, 1, [Define if the target CPU is an ARM]) ;;
+ xsparc*)
+ HAVE_CPU_SPARC=yes
+ AC_DEFINE(HAVE_CPU_SPARC, 1, [Define if the target CPU is a SPARC]) ;;
+ xmips*)
+ HAVE_CPU_MIPS=yes
+ AC_DEFINE(HAVE_CPU_MIPS, 1, [Define if the target CPU is a MIPS]) ;;
+ xhppa*)
+ HAVE_CPU_HPPA=yes
+ AC_DEFINE(HAVE_CPU_HPPA, 1, [Define if the target CPU is a HPPA]) ;;
+ xs390*)
+ HAVE_CPU_S390=yes
+ AC_DEFINE(HAVE_CPU_S390, 1, [Define if the target CPU is a S390]) ;;
+ xia64*)
+ HAVE_CPU_IA64=yes
+ AC_DEFINE(HAVE_CPU_IA64, 1, [Define if the target CPU is a IA64]) ;;
+ xm68k*)
+ HAVE_CPU_M68K=yes
+ AC_DEFINE(HAVE_CPU_M68K, 1, [Define if the target CPU is a M68K]) ;;
+ xx86_64)
+ HAVE_CPU_X86_64=yes
+ AC_DEFINE(HAVE_CPU_X86_64, 1, [Define if the target CPU is a x86_64]) ;;
+ xcris)
+ HAVE_CPU_CRIS=yes
+ AC_DEFINE(HAVE_CPU_CRIS, 1, [Define if the target CPU is a CRIS]) ;;
+ xcrisv32)
+ HAVE_CPU_CRISV32=yes
+ AC_DEFINE(HAVE_CPU_CRISV32, 1, [Define if the target CPU is a CRISv32]) ;;
+ esac
+
+ dnl Determine endianness
+ AC_C_BIGENDIAN
+
+ AM_CONDITIONAL(HAVE_CPU_I386, test "x$HAVE_CPU_I386" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_PPC, test "x$HAVE_CPU_PPC" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_PPC64, test "x$HAVE_CPU_PPC64" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_ALPHA, test "x$HAVE_CPU_ALPHA" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_ARM, test "x$HAVE_CPU_ARM" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_SPARC, test "x$HAVE_CPU_SPARC" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_HPPA, test "x$HAVE_CPU_HPPA" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_MIPS, test "x$HAVE_CPU_MIPS" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_S390, test "x$HAVE_CPU_S390" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_IA64, test "x$HAVE_CPU_IA64" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_M68K, test "x$HAVE_CPU_M68K" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_X86_64, test "x$HAVE_CPU_X86_64" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_CRIS, test "x$HAVE_CPU_CRIS" = "xyes")
+ AM_CONDITIONAL(HAVE_CPU_CRISV32, test "x$HAVE_CPU_CRISV32" = "xyes")
+
+ AC_DEFINE_UNQUOTED(HOST_CPU, "$host_cpu", [the host CPU])
+ AC_DEFINE_UNQUOTED(TARGET_CPU, "$target_cpu", [the target CPU])
+])
+
+dnl check if unaligned memory access works correctly
+AC_DEFUN([AG_GST_UNALIGNED_ACCESS], [
+ AC_MSG_CHECKING([if unaligned memory access works correctly])
+ if test x"$as_cv_unaligned_access" = x ; then
+ case $host in
+ alpha*|arm*|hp*|mips*|sh*|sparc*|ia64*)
+ _AS_ECHO_N([(blacklisted) ])
+ as_cv_unaligned_access=no
+ ;;
+ i?86*|x86_64*|amd64*|powerpc*|m68k*|cris*)
+ _AS_ECHO_N([(whitelisted) ])
+ as_cv_unaligned_access=yes
+ ;;
+ esac
+ else
+ _AS_ECHO_N([(cached) ])
+ fi
+ if test x"$as_cv_unaligned_access" = x ; then
+ AC_TRY_RUN([
+int main(int argc, char **argv)
+{
+ char array[] = "ABCDEFGH";
+ unsigned int iarray[2];
+ memcpy(iarray,array,8);
+#define GET(x) (*(unsigned int *)((char *)iarray + (x)))
+ if(GET(0) != 0x41424344 && GET(0) != 0x44434241) return 1;
+ if(GET(1) != 0x42434445 && GET(1) != 0x45444342) return 1;
+ if(GET(2) != 0x43444546 && GET(2) != 0x46454443) return 1;
+ if(GET(3) != 0x44454647 && GET(3) != 0x47464544) return 1;
+ return 0;
+}
+ ], as_cv_unaligned_access="yes", as_cv_unaligned_access="no")
+ fi
+ AC_MSG_RESULT($as_cv_unaligned_access)
+ if test "$as_cv_unaligned_access" = "yes"; then
+ AC_DEFINE_UNQUOTED(HAVE_UNALIGNED_ACCESS, 1,
+ [defined if unaligned memory access works correctly])
+ fi
+])
--- /dev/null
+dnl configure-time options shared among gstreamer modules
+
+dnl AG_GST_ARG_DEBUG
+dnl AG_GST_ARG_PROFILING
+dnl AG_GST_ARG_VALGRIND
+dnl AG_GST_ARG_GCOV
+
+dnl AG_GST_ARG_EXAMPLES
+
+dnl AG_GST_ARG_WITH_PKG_CONFIG_PATH
+dnl AG_GST_ARG_WITH_PACKAGE_NAME
+dnl AG_GST_ARG_WITH_PACKAGE_ORIGIN
+
+dnl AG_GST_ARG_WITH_PLUGINS
+dnl AG_GST_CHECK_PLUGIN
+dnl AG_GST_DISABLE_PLUGIN
+
+dnl AG_GST_ARG_ENABLE_EXTERNAL
+dnl AG_GST_ARG_ENABLE_EXPERIMENTAL
+dnl AG_GST_ARG_ENABLE_BROKEN
+
+dnl AG_GST_ARG_DISABLE_FATAL_WARNINGS
+AC_DEFUN([AG_GST_ARG_DEBUG],
+[
+ dnl debugging stuff
+ AC_ARG_ENABLE(debug,
+ AC_HELP_STRING([--disable-debug],[disable addition of -g debugging info]),
+ [
+ case "${enableval}" in
+ yes) USE_DEBUG=yes ;;
+ no) USE_DEBUG=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --enable-debug) ;;
+ esac
+ ],
+ [USE_DEBUG=yes]) dnl Default value
+])
+
+AC_DEFUN([AG_GST_ARG_PROFILING],
+[
+ AC_ARG_ENABLE(profiling,
+ AC_HELP_STRING([--enable-profiling],
+ [adds -pg to compiler commandline, for profiling]),
+ [
+ case "${enableval}" in
+ yes) USE_PROFILING=yes ;;
+ no) USE_PROFILING=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --enable-profiling) ;;
+ esac
+ ],
+ [USE_PROFILING=no]) dnl Default value
+])
+
+AC_DEFUN([AG_GST_ARG_VALGRIND],
+[
+ dnl valgrind inclusion
+ AC_ARG_ENABLE(valgrind,
+ AC_HELP_STRING([--disable-valgrind],[disable run-time valgrind detection]),
+ [
+ case "${enableval}" in
+ yes) USE_VALGRIND="$USE_DEBUG" ;;
+ no) USE_VALGRIND=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --enable-valgrind) ;;
+ esac
+ ],
+ [USE_VALGRIND="$USE_DEBUG"]) dnl Default value
+ VALGRIND_REQ="3.0"
+ if test "x$USE_VALGRIND" = xyes; then
+ PKG_CHECK_MODULES(VALGRIND, valgrind >= $VALGRIND_REQ,
+ USE_VALGRIND="yes",
+ USE_VALGRIND="no")
+ fi
+ if test "x$USE_VALGRIND" = xyes; then
+ AC_DEFINE(HAVE_VALGRIND, 1, [Define if valgrind should be used])
+ AC_MSG_NOTICE(Using extra code paths for valgrind)
+ fi
+])
+
+AC_DEFUN([AG_GST_ARG_GCOV],
+[
+ AC_ARG_ENABLE(gcov,
+ AC_HELP_STRING([--enable-gcov],
+ [compile with coverage profiling instrumentation (gcc only)]),
+ enable_gcov=$enableval,
+ enable_gcov=no)
+ if test x$enable_gcov = xyes ; then
+ if test "x$GCC" != "xyes"
+ then
+ AC_MSG_ERROR([gcov only works if gcc is used])
+ fi
+
+ AS_COMPILER_FLAG(["-fprofile-arcs"],
+ [GCOV_CFLAGS="$GCOV_CFLAGS -fprofile-arcs"],
+ true)
+ AS_COMPILER_FLAG(["-ftest-coverage"],
+ [GCOV_CFLAGS="$GCOV_CFLAGS -ftest-coverage"],
+ true)
+ dnl remove any -O flags - FIXME: is this needed ?
+ GCOV_CFLAGS=`echo "$GCOV_CFLAGS" | sed -e 's/-O[[0-9]]*//g'`
+ dnl libtool 1.5.22 and lower strip -fprofile-arcs from the flags
+ dnl passed to the linker, which is a bug; -fprofile-arcs implicitly
+ dnl links in -lgcov, so we do it explicitly here for the same effect
+ GCOV_LIBS=-lgcov
+ AC_SUBST(GCOV_CFLAGS)
+ AC_SUBST(GCOV_LIBS)
+ GCOV=`echo $CC | sed s/gcc/gcov/g`
+ AC_SUBST(GCOV)
+
+ GST_GCOV_ENABLED=yes
+ AC_DEFINE_UNQUOTED(GST_GCOV_ENABLED, 1,
+ [Defined if gcov is enabled to force a rebuild due to config.h changing])
+ dnl if gcov is used, we do not want default -O2 CFLAGS
+ if test "x$GST_GCOV_ENABLED" = "xyes"
+ then
+ CFLAGS="$CFLAGS -O0"
+ AC_SUBST(CFLAGS)
+ CXXFLAGS="$CXXFLAGS -O0"
+ AC_SUBST(CXXFLAGS)
+ FFLAGS="$FFLAGS -O0"
+ AC_SUBST(FFLAGS)
+ CCASFLAGS="$CCASFLAGS -O0"
+ AC_SUBST(CCASFLAGS)
+ AC_MSG_NOTICE([gcov enabled, setting CFLAGS and friends to $CFLAGS])
+ fi
+ fi
+ AM_CONDITIONAL(GST_GCOV_ENABLED, test x$enable_gcov = xyes)
+])
+
+AC_DEFUN([AG_GST_ARG_EXAMPLES],
+[
+ AC_ARG_ENABLE(examples,
+ AC_HELP_STRING([--disable-examples], [disable building examples]),
+ [
+ case "${enableval}" in
+ yes) BUILD_EXAMPLES=yes ;;
+ no) BUILD_EXAMPLES=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --disable-examples) ;;
+ esac
+ ],
+ [BUILD_EXAMPLES=yes]) dnl Default value
+ AM_CONDITIONAL(BUILD_EXAMPLES, test "x$BUILD_EXAMPLES" = "xyes")
+])
+
+AC_DEFUN([AG_GST_ARG_WITH_PKG_CONFIG_PATH],
+[
+ dnl possibly modify pkg-config path
+ AC_ARG_WITH(pkg-config-path,
+ AC_HELP_STRING([--with-pkg-config-path],
+ [colon-separated list of pkg-config(1) dirs]),
+ [
+ export PKG_CONFIG_PATH=${withval}
+ AC_MSG_NOTICE(Set PKG_CONFIG_PATH to $PKG_CONFIG_PATH)
+ ])
+])
+
+
+dnl This macro requires that GST_GIT or GST_CVS is set to yes or no (release)
+AC_DEFUN([AG_GST_ARG_WITH_PACKAGE_NAME],
+[
+ dnl package name in plugins
+ AC_ARG_WITH(package-name,
+ AC_HELP_STRING([--with-package-name],
+ [specify package name to use in plugins]),
+ [
+ case "${withval}" in
+ yes) AC_MSG_ERROR(bad value ${withval} for --with-package-name) ;;
+ no) AC_MSG_ERROR(bad value ${withval} for --with-package-name) ;;
+ *) GST_PACKAGE_NAME="${withval}" ;;
+ esac
+ ],
+ [
+ P=$1
+ if test "x$P" = "x"
+ then
+ P=$PACKAGE_NAME
+ fi
+
+ if test "x$PACKAGE_VERSION_NANO" = "x0"
+ then
+ GST_PACKAGE_NAME="$P source release"
+ else
+ if test "x$PACKAGE_VERSION_NANO" = "x1"
+ then
+ GST_PACKAGE_NAME="$P git"
+ else
+ GST_PACKAGE_NAME="$P prerelease"
+ fi
+ fi
+ ]
+ )
+ AC_MSG_NOTICE(Using $GST_PACKAGE_NAME as package name)
+ AC_DEFINE_UNQUOTED(GST_PACKAGE_NAME, "$GST_PACKAGE_NAME",
+ [package name in plugins])
+ AC_SUBST(GST_PACKAGE_NAME)
+])
+
+AC_DEFUN([AG_GST_ARG_WITH_PACKAGE_ORIGIN],
+[
+ dnl package origin URL
+ AC_ARG_WITH(package-origin,
+ AC_HELP_STRING([--with-package-origin],
+ [specify package origin URL to use in plugins]),
+ [
+ case "${withval}" in
+ yes) AC_MSG_ERROR(bad value ${withval} for --with-package-origin) ;;
+ no) AC_MSG_ERROR(bad value ${withval} for --with-package-origin) ;;
+ *) GST_PACKAGE_ORIGIN="${withval}" ;;
+ esac
+ ],
+ [GST_PACKAGE_ORIGIN="[Unknown package origin]"] dnl Default value
+ )
+ AC_MSG_NOTICE(Using $GST_PACKAGE_ORIGIN as package origin)
+ AC_DEFINE_UNQUOTED(GST_PACKAGE_ORIGIN, "$GST_PACKAGE_ORIGIN",
+ [package origin])
+ AC_SUBST(GST_PACKAGE_ORIGIN)
+])
+
+dnl sets WITH_PLUGINS to the list of plug-ins given as an argument
+dnl also clears GST_PLUGINS_ALL and GST_PLUGINS_SELECTED
+AC_DEFUN([AG_GST_ARG_WITH_PLUGINS],
+[
+ AC_ARG_WITH(plugins,
+ AC_HELP_STRING([--with-plugins],
+ [comma-separated list of dependencyless plug-ins to compile]),
+ [WITH_PLUGINS=$withval],
+ [WITH_PLUGINS=])
+
+ GST_PLUGINS_ALL=""
+ GST_PLUGINS_SELECTED=""
+ GST_PLUGINS_NONPORTED=""
+
+ AC_SUBST(GST_PLUGINS_ALL)
+ AC_SUBST(GST_PLUGINS_SELECTED)
+ AC_SUBST(GST_PLUGINS_NONPORTED)
+])
+
+dnl AG_GST_CHECK_PLUGIN(PLUGIN-NAME)
+dnl
+dnl This macro adds the plug-in <PLUGIN-NAME> to GST_PLUGINS_ALL. Then it
+dnl checks if WITH_PLUGINS is empty or the plugin is present in WITH_PLUGINS,
+dnl and if so adds it to GST_PLUGINS_SELECTED. Then it checks if the plugin
+dnl is present in WITHOUT_PLUGINS (ie. was disabled specifically) and if so
+dnl removes it from GST_PLUGINS_SELECTED.
+dnl
+dnl The macro will call AM_CONDITIONAL(USE_PLUGIN_<PLUGIN-NAME>, ...) to allow
+dnl control of what is built in Makefile.ams.
+AC_DEFUN([AG_GST_CHECK_PLUGIN],
+[
+ GST_PLUGINS_ALL="$GST_PLUGINS_ALL [$1]"
+
+ define([pname_def],translit([$1], -a-z, _a-z))
+
+ AC_ARG_ENABLE([$1],
+ AC_HELP_STRING([--disable-[$1]], [disable dependency-less $1 plugin]),
+ [
+ case "${enableval}" in
+ yes) [gst_use_]pname_def=yes ;;
+ no) [gst_use_]pname_def=no ;;
+ *) AC_MSG_ERROR([bad value ${enableval} for --enable-$1]) ;;
+ esac
+ ],
+ [[gst_use_]pname_def=yes]) dnl Default value
+
+ if test x$[gst_use_]pname_def = xno; then
+ AC_MSG_NOTICE(disabling dependency-less plugin $1)
+ WITHOUT_PLUGINS="$WITHOUT_PLUGINS [$1]"
+ fi
+ undefine([pname_def])
+
+ dnl First check inclusion
+ if [[ -z "$WITH_PLUGINS" ]] || echo " [$WITH_PLUGINS] " | tr , ' ' | grep -i " [$1] " > /dev/null; then
+ GST_PLUGINS_SELECTED="$GST_PLUGINS_SELECTED [$1]"
+ fi
+ dnl Then check exclusion
+ if echo " [$WITHOUT_PLUGINS] " | tr , ' ' | grep -i " [$1] " > /dev/null; then
+ GST_PLUGINS_SELECTED=`echo " $GST_PLUGINS_SELECTED " | $SED -e 's/ [$1] / /'`
+ fi
+ dnl Finally check if the plugin is ported or not
+ if echo " [$GST_PLUGINS_NONPORTED] " | tr , ' ' | grep -i " [$1] " > /dev/null; then
+ GST_PLUGINS_SELECTED=`echo " $GST_PLUGINS_SELECTED " | $SED -e 's/ [$1] / /'`
+ fi
+ AM_CONDITIONAL([USE_PLUGIN_]translit([$1], a-z, A-Z), echo " $GST_PLUGINS_SELECTED " | grep -i " [$1] " > /dev/null)
+])
+
+dnl AG_GST_DISABLE_PLUGIN(PLUGIN-NAME)
+dnl
+dnl This macro disables the plug-in <PLUGIN-NAME> by removing it from
+dnl GST_PLUGINS_SELECTED.
+AC_DEFUN([AG_GST_DISABLE_PLUGIN],
+[
+ GST_PLUGINS_SELECTED=`echo " $GST_PLUGINS_SELECTED " | $SED -e 's/ [$1] / /'`
+ AM_CONDITIONAL([USE_PLUGIN_]translit([$1], a-z, A-Z), false)
+])
+
+AC_DEFUN([AG_GST_ARG_ENABLE_EXTERNAL],
+[
+ AG_GST_CHECK_FEATURE(EXTERNAL, [building of plug-ins with external deps],,
+ HAVE_EXTERNAL=yes, enabled,
+ [
+ AC_MSG_NOTICE(building external plug-ins)
+ BUILD_EXTERNAL="yes"
+ ],[
+ AC_MSG_WARN(all plug-ins with external dependencies will not be built)
+ BUILD_EXTERNAL="no"
+ ])
+ # make BUILD_EXTERNAL available to Makefile.am
+ AM_CONDITIONAL(BUILD_EXTERNAL, test "x$BUILD_EXTERNAL" = "xyes")
+])
+
+dnl experimental plug-ins; stuff that hasn't had the dust settle yet
+dnl read 'builds, but might not work'
+AC_DEFUN([AG_GST_ARG_ENABLE_EXPERIMENTAL],
+[
+ AG_GST_CHECK_FEATURE(EXPERIMENTAL, [building of experimental plug-ins],,
+ HAVE_EXPERIMENTAL=yes, disabled,
+ [
+ AC_MSG_WARN(building experimental plug-ins)
+ BUILD_EXPERIMENTAL="yes"
+ ],[
+ AC_MSG_NOTICE(not building experimental plug-ins)
+ BUILD_EXPERIMENTAL="no"
+ ])
+ # make BUILD_EXPERIMENTAL available to Makefile.am
+ AM_CONDITIONAL(BUILD_EXPERIMENTAL, test "x$BUILD_EXPERIMENTAL" = "xyes")
+])
+
+dnl broken plug-ins; stuff that doesn't seem to build at the moment
+AC_DEFUN([AG_GST_ARG_ENABLE_BROKEN],
+[
+ AG_GST_CHECK_FEATURE(BROKEN, [building of broken plug-ins],,
+ HAVE_BROKEN=yes, disabled,
+ [
+ AC_MSG_WARN([building broken plug-ins -- no bug reports on these, only patches ...])
+ ],[
+ AC_MSG_NOTICE([not building broken plug-ins])
+ ])
+])
+
+dnl allow people (or build tools) to override default behaviour
+dnl for fatal compiler warnings
+dnl Enable fatal warnings by default only for development versions
+AC_DEFUN([AG_GST_ARG_DISABLE_FATAL_WARNINGS],
+[
+ AC_ARG_ENABLE(fatal-warnings,
+ AC_HELP_STRING([--disable-fatal-warnings],
+ [Don't turn compiler warnings into fatal errors]),
+ [
+ case "${enableval}" in
+ yes) FATAL_WARNINGS=yes ;;
+ no) FATAL_WARNINGS=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --disable-fatal-warnings) ;;
+ esac
+ ],
+ [
+ if test "x`expr $PACKAGE_VERSION_MINOR % 2`" = "x1" -a "x`expr $PACKAGE_VERSION_MICRO '<' 90`" = "x1"; then
+ FATAL_WARNINGS=yes
+ else
+ FATAL_WARNINGS=no
+ fi
+ ])
+])
--- /dev/null
+dnl pkg-config-based checks for GStreamer modules and dependency modules
+
+dnl generic:
+dnl AG_GST_PKG_CHECK_MODULES([PREFIX], [WHICH], [REQUIRED])
+dnl sets HAVE_[$PREFIX], [$PREFIX]_*
+dnl AG_GST_CHECK_MODULES([PREFIX], [MODULE], [MINVER], [NAME], [REQUIRED])
+dnl sets HAVE_[$PREFIX], [$PREFIX]_*
+
+dnl specific:
+dnl AG_GST_CHECK_GST([MAJMIN], [MINVER], [REQUIRED])
+dnl also sets/ACSUBSTs GST_TOOLS_DIR and GST_PLUGINS_DIR
+dnl AG_GST_CHECK_GST_BASE([MAJMIN], [MINVER], [REQUIRED])
+dnl AG_GST_CHECK_GST_CONTROLLER([MAJMIN], [MINVER], [REQUIRED])
+dnl AG_GST_CHECK_GST_NET([MAJMIN], [MINVER], [REQUIRED])
+dnl AG_GST_CHECK_GST_CHECK([MAJMIN], [MINVER], [REQUIRED])
+dnl AG_GST_CHECK_GST_PLUGINS_BASE([MAJMIN], [MINVER], [REQUIRED])
+dnl also sets/ACSUBSTs GSTPB_PLUGINS_DIR
+
+AC_DEFUN([AG_GST_PKG_CHECK_MODULES],
+[
+ which="[$2]"
+ dnl not required by default, since we use this mostly for plugin deps
+ required=ifelse([$3], , "no", [$3])
+
+ PKG_CHECK_MODULES([$1], $which,
+ [
+ HAVE_[$1]="yes"
+ ],
+ [
+ HAVE_[$1]="no"
+ if test "x$required" = "xyes"; then
+ AC_MSG_ERROR($[$1]_PKG_ERRORS)
+ else
+ AC_MSG_NOTICE($[$1]_PKG_ERRORS)
+ fi
+ ])
+
+ dnl AC_SUBST of CFLAGS and LIBS was not done before automake 1.7
+ dnl It gets done automatically in automake >= 1.7, which we now require
+]))
+
+AC_DEFUN([AG_GST_CHECK_MODULES],
+[
+ module=[$2]
+ minver=[$3]
+ name="[$4]"
+ required=ifelse([$5], , "yes", [$5]) dnl required by default
+
+ PKG_CHECK_MODULES([$1], $module >= $minver,
+ [
+ HAVE_[$1]="yes"
+ ],
+ [
+ HAVE_[$1]="no"
+ AC_MSG_NOTICE($[$1]_PKG_ERRORS)
+ if test "x$required" = "xyes"; then
+ AC_MSG_ERROR([no $module >= $minver ($name) found])
+ else
+ AC_MSG_NOTICE([no $module >= $minver ($name) found])
+ fi
+ ])
+
+ dnl AC_SUBST of CFLAGS and LIBS was not done before automake 1.7
+ dnl It gets done automatically in automake >= 1.7, which we now require
+]))
+
+AC_DEFUN([AG_GST_CHECK_GST],
+[
+ AG_GST_CHECK_MODULES(GST, gstreamer-[$1], [$2], [GStreamer], [$3])
+ dnl allow setting before calling this macro to override
+ if test -z $GST_TOOLS_DIR; then
+ GST_TOOLS_DIR=`$PKG_CONFIG --variable=toolsdir gstreamer-[$1]`
+ if test -z $GST_TOOLS_DIR; then
+ AC_MSG_ERROR(
+ [no tools dir set in GStreamer pkg-config file, core upgrade needed.])
+ fi
+ fi
+ AC_MSG_NOTICE([using GStreamer tools in $GST_TOOLS_DIR])
+ AC_SUBST(GST_TOOLS_DIR)
+
+ dnl check for where core plug-ins got installed
+ dnl this is used for unit tests
+ dnl allow setting before calling this macro to override
+ if test -z $GST_PLUGINS_DIR; then
+ GST_PLUGINS_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-[$1]`
+ if test -z $GST_PLUGINS_DIR; then
+ AC_MSG_ERROR(
+ [no pluginsdir set in GStreamer pkg-config file, core upgrade needed.])
+ fi
+ fi
+ AC_MSG_NOTICE([using GStreamer plug-ins in $GST_PLUGINS_DIR])
+ AC_SUBST(GST_PLUGINS_DIR)
+])
+
+AC_DEFUN([AG_GST_CHECK_GST_BASE],
+[
+ AG_GST_CHECK_MODULES(GST_BASE, gstreamer-base-[$1], [$2],
+ [GStreamer Base Libraries], [$3])
+])
+
+AC_DEFUN([AG_GST_CHECK_GST_CONTROLLER],
+[
+ AG_GST_CHECK_MODULES(GST_CONTROLLER, gstreamer-controller-[$1], [$2],
+ [GStreamer Controller Library], [$3])
+])
+
+AC_DEFUN([AG_GST_CHECK_GST_NET],
+[
+ AG_GST_CHECK_MODULES(GST_NET, gstreamer-net-[$1], [$2],
+ [GStreamer Network Library], [$3])
+])
+
+AC_DEFUN([AG_GST_CHECK_GST_CHECK],
+[
+ AG_GST_CHECK_MODULES(GST_CHECK, gstreamer-check-[$1], [$2],
+ [GStreamer Check unittest Library], [$3])
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_UNINSTALLED_SETUP([ACTION-IF-UNINSTALLED], [ACTION-IF-NOT])
+dnl
+dnl ACTION-IF-UNINSTALLED (optional) extra actions to perform if the setup
+dnl is an uninstalled setup
+dnl ACTION-IF-NOT (optional) extra actions to perform if the setup
+dnl is not an uninstalled setup
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_UNINSTALLED_SETUP],
+[
+ AC_MSG_CHECKING([whether this is an uninstalled GStreamer setup])
+ AC_CACHE_VAL(gst_cv_is_uninstalled_setup,[
+ gst_cv_is_uninstalled_setup=no
+ if (set -u; : $GST_PLUGIN_SYSTEM_PATH) 2>/dev/null ; then
+ if test -z "$GST_PLUGIN_SYSTEM_PATH" \
+ -a -n "$GST_PLUGIN_SCANNER" \
+ -a -n "$GST_PLUGIN_PATH" \
+ -a -n "$GST_REGISTRY" \
+ -a -n "$DYLD_LIBRARY_PATH" \
+ -a -n "$LD_LIBRARY_PATH"; then
+ gst_cv_is_uninstalled_setup=yes;
+ fi
+ fi
+ ])
+ AC_MSG_RESULT($gst_cv_is_uninstalled_setup)
+ if test "x$gst_cv_is_uninstalled_setup" = "xyes"; then
+ ifelse([$1], , :, [$1])
+ else
+ ifelse([$2], , :, [$2])
+ fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_GST_PLUGINS_BASE([GST-API_VERSION], [MIN-VERSION], [REQUIRED])
+dnl
+dnl Sets GST_PLUGINS_BASE_CFLAGS and GST_PLUGINS_BASE_LIBS.
+dnl
+dnl Also sets GSTPB_PLUGINS_DIR (and for consistency also GST_PLUGINS_BASE_DIR)
+dnl for use in Makefile.am. This is only really needed/useful in uninstalled
+dnl setups, since in an installed setup all plugins will be found in
+dnl GST_PLUGINS_DIR anyway.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_GST_PLUGINS_BASE],
+[
+ AG_GST_CHECK_MODULES(GST_PLUGINS_BASE, gstreamer-plugins-base-[$1], [$2],
+ [GStreamer Base Plugins], [$3])
+
+ if test "x$HAVE_GST_PLUGINS_BASE" = "xyes"; then
+ dnl check for where base plugins got installed
+ dnl this is used for unit tests
+ dnl allow setting before calling this macro to override
+ if test -z $GSTPB_PLUGINS_DIR; then
+ GSTPB_PLUGINS_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-plugins-base-[$1]`
+ if test -z $GSTPB_PLUGINS_DIR; then
+ AC_MSG_ERROR(
+ [no pluginsdir set in GStreamer Base Plugins pkg-config file])
+ fi
+ fi
+ AC_MSG_NOTICE([using GStreamer Base Plugins in $GSTPB_PLUGINS_DIR])
+ GST_PLUGINS_BASE_DIR="$GSTPB_PLUGINS_DIR/gst:$GSTPB_PLUGINS_DIR/sys:$GSTPB_PLUGINS_DIR/ext"
+ AC_SUBST(GST_PLUGINS_BASE_DIR)
+ AC_SUBST(GSTPB_PLUGINS_DIR)
+ fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_GST_PLUGINS_GOOD([GST-API_VERSION], [MIN-VERSION])
+dnl
+dnl Will set GST_PLUGINS_GOOD_DIR for use in Makefile.am. Note that this will
+dnl only be set in an uninstalled setup, since -good ships no .pc file and in
+dnl an installed setup all plugins will be found in GST_PLUGINS_DIR anyway.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_GST_PLUGINS_GOOD],
+[
+ AG_GST_CHECK_MODULES(GST_PLUGINS_GOOD, gstreamer-plugins-good-[$1], [$2],
+ [GStreamer Good Plugins], [no])
+
+ if test "x$HAVE_GST_PLUGINS_GOOD" = "xyes"; then
+ dnl check for where good plugins got installed
+ dnl this is used for unit tests
+ dnl allow setting before calling this macro to override
+ if test -z $GST_PLUGINS_GOOD_DIR; then
+ GST_PLUGINS_GOOD_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-plugins-good-[$1]`
+ if test -z $GST_PLUGINS_GOOD_DIR; then
+ AC_MSG_ERROR([no pluginsdir set in GStreamer Good Plugins pkg-config file])
+ fi
+ fi
+ AC_MSG_NOTICE([using GStreamer Good Plugins in $GST_PLUGINS_GOOD_DIR])
+ GST_PLUGINS_GOOD_DIR="$GST_PLUGINS_GOOD_DIR/gst:$GST_PLUGINS_GOOD_DIR/sys:$GST_PLUGINS_GOOD_DIR/ext"
+ AC_SUBST(GST_PLUGINS_GOOD_DIR)
+ fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_GST_PLUGINS_UGLY([GST-API_VERSION], [MIN-VERSION])
+dnl
+dnl Will set GST_PLUGINS_UGLY_DIR for use in Makefile.am. Note that this will
+dnl only be set in an uninstalled setup, since -bad ships no .pc file and in
+dnl an installed setup all plugins will be found in GST_PLUGINS_DIR anyway.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_GST_PLUGINS_UGLY],
+[
+ AG_GST_CHECK_MODULES(GST_PLUGINS_UGLY, gstreamer-plugins-ugly-[$1], [$2],
+ [GStreamer Ugly Plugins], [no])
+
+ if test "x$HAVE_GST_PLUGINS_UGLY" = "xyes"; then
+ dnl check for where ugly plugins got installed
+ dnl this is used for unit tests
+ dnl allow setting before calling this macro to override
+ if test -z $GST_PLUGINS_UGLY_DIR; then
+ GST_PLUGINS_UGLY_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-plugins-ugly-[$1]`
+ if test -z $GST_PLUGINS_UGLY_DIR; then
+ AC_MSG_ERROR([no pluginsdir set in GStreamer Ugly Plugins pkg-config file])
+ fi
+ fi
+ AC_MSG_NOTICE([using GStreamer Ugly Plugins in $GST_PLUGINS_UGLY_DIR])
+ GST_PLUGINS_UGLY_DIR="$GST_PLUGINS_UGLY_DIR/gst:$GST_PLUGINS_UGLY_DIR/sys:$GST_PLUGINS_UGLY_DIR/ext"
+ AC_SUBST(GST_PLUGINS_UGLY_DIR)
+ fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_GST_PLUGINS_BAD([GST-API_VERSION], [MIN-VERSION])
+dnl
+dnl Will set GST_PLUGINS_BAD_DIR for use in Makefile.am. Note that this will
+dnl only be set in an uninstalled setup, since -ugly ships no .pc file and in
+dnl an installed setup all plugins will be found in GST_PLUGINS_DIR anyway.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_GST_PLUGINS_BAD],
+[
+ AG_GST_CHECK_MODULES(GST_PLUGINS_BAD, gstreamer-plugins-bad-[$1], [$2],
+ [GStreamer Bad Plugins], [no])
+
+ if test "x$HAVE_GST_PLUGINS_BAD" = "xyes"; then
+ dnl check for where bad plugins got installed
+ dnl this is used for unit tests
+ dnl allow setting before calling this macro to override
+ if test -z $GST_PLUGINS_BAD_DIR; then
+ GST_PLUGINS_BAD_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-plugins-bad-[$1]`
+ if test -z $GST_PLUGINS_BAD_DIR; then
+ AC_MSG_ERROR([no pluginsdir set in GStreamer Bad Plugins pkg-config file])
+ fi
+ fi
+ AC_MSG_NOTICE([using GStreamer Bad Plugins in $GST_PLUGINS_BAD_DIR])
+ GST_PLUGINS_BAD_DIR="$GST_PLUGINS_BAD_DIR/gst:$GST_PLUGINS_BAD_DIR/sys:$GST_PLUGINS_BAD_DIR/ext"
+ AC_SUBST(GST_PLUGINS_BAD_DIR)
+ fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_GST_PLUGINS_LIBAV([GST-API_VERSION], [MIN-VERSION])
+dnl
+dnl Will set GST_PLUGINS_LIBAV_DIR for use in Makefile.am. Note that this will
+dnl only be set in an uninstalled setup, since -libav ships no .pc file and in
+dnl an installed setup all plugins will be found in GST_PLUGINS_DIR anyway.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_GST_PLUGINS_LIBAV],
+[
+ AG_GST_CHECK_MODULES(GST_PLUGINS_LIBAV, gstreamer-plugins-libav-[$1], [$2],
+ [GStreamer Libav Plugins], [no])
+
+ if test "x$HAVE_GST_PLUGINS_LIBAV" = "xyes"; then
+ dnl check for where libav plugins got installed
+ dnl this is used for unit tests
+ dnl allow setting before calling this macro to override
+ if test -z $GST_PLUGINS_LIBAV_DIR; then
+ GST_PLUGINS_LIBAV_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-plugins-libav-[$1]`
+ if test -z $GST_PLUGINS_LIBAV_DIR; then
+ AC_MSG_ERROR([no pluginsdir set in GStreamer Libav Plugins pkg-config file])
+ fi
+ fi
+ GST_PLUGINS_LIBAV_DIR="$GST_PLUGINS_LIBAV_DIR/ext/libav"
+ AC_MSG_NOTICE([using GStreamer Libav Plugins in $GST_PLUGINS_LIBAV_DIR])
+ AC_SUBST(GST_PLUGINS_LIBAV_DIR)
+ fi
+])
--- /dev/null
+AC_DEFUN([AG_GST_DEBUGINFO], [
+AC_ARG_ENABLE(debug,
+AC_HELP_STRING([--disable-debug],[disable addition of -g debugging info]),
+[case "${enableval}" in
+ yes) USE_DEBUG=yes ;;
+ no) USE_DEBUG=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --enable-debug) ;;
+esac],
+[USE_DEBUG=yes]) dnl Default value
+
+AC_ARG_ENABLE(DEBUG,
+AC_HELP_STRING([--disable-DEBUG],[disables compilation of debugging messages]),
+[case "${enableval}" in
+ yes) ENABLE_DEBUG=yes ;;
+ no) ENABLE_DEBUG=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --enable-DEBUG) ;;
+esac],
+[ENABLE_DEBUG=yes]) dnl Default value
+if test x$ENABLE_DEBUG = xyes; then
+ AC_DEFINE(GST_DEBUG_ENABLED, 1, [Define if DEBUG statements should be compiled in])
+fi
+
+AC_ARG_ENABLE(INFO,
+AC_HELP_STRING([--disable-INFO],[disables compilation of informational messages]),
+[case "${enableval}" in
+ yes) ENABLE_INFO=yes ;;
+ no) ENABLE_INFO=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --enable-INFO) ;;
+esac],
+[ENABLE_INFO=yes]) dnl Default value
+if test x$ENABLE_INFO = xyes; then
+ AC_DEFINE(GST_INFO_ENABLED, 1, [Define if INFO statements should be compiled in])
+fi
+
+AC_ARG_ENABLE(debug-color,
+AC_HELP_STRING([--disable-debug-color],[disables color output of DEBUG and INFO output]),
+[case "${enableval}" in
+ yes) ENABLE_DEBUG_COLOR=yes ;;
+ no) ENABLE_DEBUG_COLOR=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --enable-debug-color) ;;
+esac],
+[ENABLE_DEBUG_COLOR=yes]) dnl Default value
+if test "x$ENABLE_DEBUG_COLOR" = xyes; then
+ AC_DEFINE(GST_DEBUG_COLOR, 1, [Define if debugging messages should be colorized])
+fi
+])
--- /dev/null
+dnl default elements used for tests and such
+
+dnl AG_GST_DEFAULT_ELEMENTS
+
+AC_DEFUN([AG_GST_DEFAULT_ELEMENTS],
+[
+ dnl decide on default elements
+ dnl FIXME: describe where exactly this gets used
+ dnl FIXME: decide if it's a problem that this could point to sinks from
+ dnl depending plugin modules
+ dnl FIXME: when can we just use autoaudiosrc and autovideosrc?
+ DEFAULT_AUDIOSINK="autoaudiosink"
+ DEFAULT_VIDEOSINK="autovideosink"
+ DEFAULT_AUDIOSRC="alsasrc"
+ DEFAULT_VIDEOSRC="v4l2src"
+ DEFAULT_VISUALIZER="goom"
+ case "$host" in
+ *-sun-* | *pc-solaris* )
+ DEFAULT_AUDIOSRC="sunaudiosrc"
+ ;;
+ *-darwin* )
+ DEFAULT_AUDIOSRC="osxaudiosrc"
+ ;;
+ esac
+
+ dnl Default audio sink
+ AC_ARG_WITH(default-audiosink,
+ AC_HELP_STRING([--with-default-audiosink], [specify default audio sink]),
+ [
+ case "${withval}" in
+ yes) AC_MSG_ERROR(bad value ${withval} for --with-default-audiosink) ;;
+ no) AC_MSG_ERROR(bad value ${withval} for --with-default-audiosink) ;;
+ *) DEFAULT_AUDIOSINK="${withval}" ;;
+ esac
+ ],
+ [
+ DEFAULT_AUDIOSINK="$DEFAULT_AUDIOSINK"
+ ] dnl Default value as determined above
+ )
+ AC_MSG_NOTICE(Using $DEFAULT_AUDIOSINK as default audio sink)
+ AC_SUBST(DEFAULT_AUDIOSINK)
+ AC_DEFINE_UNQUOTED(DEFAULT_AUDIOSINK, "$DEFAULT_AUDIOSINK",
+ [Default audio sink])
+
+ dnl Default audio source
+ AC_ARG_WITH(default-audiosrc,
+ AC_HELP_STRING([--with-default-audiosrc], [specify default audio source]),
+ [
+ case "${withval}" in
+ yes) AC_MSG_ERROR(bad value ${withval} for --with-default-audiosrc) ;;
+ no) AC_MSG_ERROR(bad value ${withval} for --with-default-audiosrc) ;;
+ *) DEFAULT_AUDIOSRC="${withval}" ;;
+ esac
+ ],
+ [
+ DEFAULT_AUDIOSRC="$DEFAULT_AUDIOSRC"
+ ] dnl Default value as determined above
+ )
+ AC_MSG_NOTICE(Using $DEFAULT_AUDIOSRC as default audio source)
+ AC_SUBST(DEFAULT_AUDIOSRC)
+ AC_DEFINE_UNQUOTED(DEFAULT_AUDIOSRC, "$DEFAULT_AUDIOSRC",
+ [Default audio source])
+
+ dnl Default video sink
+ AC_ARG_WITH(default-videosink,
+ AC_HELP_STRING([--with-default-videosink], [specify default video sink]),
+ [
+ case "${withval}" in
+ yes) AC_MSG_ERROR(bad value ${withval} for --with-default-videosink) ;;
+ no) AC_MSG_ERROR(bad value ${withval} for --with-default-videosink) ;;
+ *) DEFAULT_VIDEOSINK="${withval}" ;;
+ esac
+ ],
+ [
+ DEFAULT_VIDEOSINK="$DEFAULT_VIDEOSINK"
+ ] dnl Default value as determined above
+ )
+ AC_MSG_NOTICE(Using $DEFAULT_VIDEOSINK as default video sink)
+ AC_SUBST(DEFAULT_VIDEOSINK)
+ AC_DEFINE_UNQUOTED(DEFAULT_VIDEOSINK, "$DEFAULT_VIDEOSINK",
+ [Default video sink])
+
+ dnl Default video source
+ AC_ARG_WITH(default-videosrc,
+ AC_HELP_STRING([--with-default-videosrc], [specify default video source]),
+ [
+ case "${withval}" in
+ yes) AC_MSG_ERROR(bad value ${withval} for --with-default-videosrc) ;;
+ no) AC_MSG_ERROR(bad value ${withval} for --with-default-videosrc) ;;
+ *) DEFAULT_VIDEOSRC="${withval}" ;;
+ esac
+ ],
+ [
+ DEFAULT_VIDEOSRC="$DEFAULT_VIDEOSRC"
+ ] dnl Default value as determined above
+ )
+ AC_MSG_NOTICE(Using $DEFAULT_VIDEOSRC as default video source)
+ AC_SUBST(DEFAULT_VIDEOSRC)
+ AC_DEFINE_UNQUOTED(DEFAULT_VIDEOSRC, "$DEFAULT_VIDEOSRC",
+ [Default video source])
+
+ dnl Default visualizer
+ AC_ARG_WITH(default-visualizer,
+ AC_HELP_STRING([--with-default-visualizer], [specify default visualizer]),
+ [
+ case "${withval}" in
+ yes) AC_MSG_ERROR(bad value ${withval} for --with-default-visualizer) ;;
+ no) AC_MSG_ERROR(bad value ${withval} for --with-default-visualizer) ;;
+ *) DEFAULT_VISUALIZER="${withval}" ;;
+ esac
+ ],
+ [
+ DEFAULT_VISUALIZER="$DEFAULT_VISUALIZER"
+ ] dnl Default value as determined above
+ )
+ AC_MSG_NOTICE(Using $DEFAULT_VISUALIZER as default visualizer)
+ AC_SUBST(DEFAULT_VISUALIZER)
+ AC_DEFINE_UNQUOTED(DEFAULT_VISUALIZER, "$DEFAULT_VISUALIZER",
+ [Default visualizer])
+])
--- /dev/null
+AC_DEFUN([AG_GST_DOCBOOK_CHECK],
+[
+ dnl choose a location to install docbook docs in
+ if test "x$PACKAGE_TARNAME" = "x"
+ then
+ AC_MSG_ERROR([Internal error - PACKAGE_TARNAME not set])
+ fi
+ docdir="\$(datadir)/doc/$PACKAGE_TARNAME-$GST_API_VERSION"
+
+ dnl enable/disable docbook documentation building
+ AC_ARG_ENABLE(docbook,
+ AC_HELP_STRING([--enable-docbook],
+ [use docbook to build documentation [default=no]]),,
+ enable_docbook=no)
+
+ have_docbook=no
+
+ if test x$enable_docbook = xyes; then
+ dnl check if we actually have everything we need
+
+ dnl check for docbook tools
+ AC_CHECK_PROG(HAVE_DOCBOOK2PS, docbook2ps, yes, no)
+ AC_CHECK_PROG(HAVE_XSLTPROC, xsltproc, yes, no)
+ AC_CHECK_PROG(HAVE_JADETEX, jadetex, yes, no)
+ AC_CHECK_PROG(HAVE_PS2PDF, ps2pdf, yes, no)
+
+ dnl check if we can process docbook stuff
+ AS_DOCBOOK(have_docbook=yes, have_docbook=no)
+
+ dnl check for extra tools
+ AC_CHECK_PROG(HAVE_DVIPS, dvips, yes, no)
+ AC_CHECK_PROG(HAVE_XMLLINT, xmllint, yes, no)
+
+ AC_CHECK_PROG(HAVE_PNGTOPNM, pngtopnm, yes, no)
+ AC_CHECK_PROG(HAVE_PNMTOPS, pnmtops, yes, no)
+ AC_CHECK_PROG(HAVE_EPSTOPDF, epstopdf, yes, no)
+
+ dnl check if we can generate HTML
+ if test "x$HAVE_XSLTPROC" = "xyes" && \
+ test "x$enable_docbook" = "xyes" && \
+ test "x$HAVE_XMLLINT" = "xyes"; then
+ DOC_HTML=yes
+ AC_MSG_NOTICE(Will output HTML documentation)
+ else
+ DOC_HTML=no
+ AC_MSG_NOTICE(Will not output HTML documentation)
+ fi
+
+ dnl check if we can generate PS
+ if test "x$HAVE_DOCBOOK2PS" = "xyes" && \
+ test "x$enable_docbook" = "xyes" && \
+ test "x$HAVE_XMLLINT" = "xyes" && \
+ test "x$HAVE_JADETEX" = "xyes" && \
+ test "x$HAVE_DVIPS" = "xyes" && \
+ test "x$HAVE_PNGTOPNM" = "xyes" && \
+ test "x$HAVE_PNMTOPS" = "xyes"; then
+ DOC_PS=yes
+ AC_MSG_NOTICE(Will output PS documentation)
+ else
+ DOC_PS=no
+ AC_MSG_NOTICE(Will not output PS documentation)
+ fi
+
+ dnl check if we can generate PDF - using only ps2pdf
+ if test "x$DOC_PS" = "xyes" && \
+ test "x$enable_docbook" = "xyes" && \
+ test "x$HAVE_XMLLINT" = "xyes" && \
+ test "x$HAVE_PS2PDF" = "xyes"; then
+ DOC_PDF=yes
+ AC_MSG_NOTICE(Will output PDF documentation)
+ else
+ DOC_PDF=no
+ AC_MSG_NOTICE(Will not output PDF documentation)
+ fi
+
+ dnl if we don't have everything, we should disable
+ if test "x$have_docbook" != "xyes"; then
+ enable_docbook=no
+ fi
+ fi
+
+ dnl if we're going to install documentation, tell us where
+ if test "x$have_docbook" = "xyes"; then
+ AC_MSG_NOTICE(Installing documentation in $docdir)
+ AC_SUBST(docdir)
+ fi
+
+ AM_CONDITIONAL(ENABLE_DOCBOOK, test x$enable_docbook = xyes)
+ AM_CONDITIONAL(DOC_HTML, test x$DOC_HTML = xyes)
+ AM_CONDITIONAL(DOC_PDF, test x$DOC_PDF = xyes)
+ AM_CONDITIONAL(DOC_PS, test x$DOC_PS = xyes)
+])
--- /dev/null
+dnl
+dnl Check for working do while(0) macros. This is used by G_STMT_START
+dnl and G_STMT_END in glib/gmacros.h. Without having this defined we
+dnl get "ambigious if-else" compiler warnings when compling C++ code.
+dnl
+dnl Copied from GLib's configure.in
+dnl
+AC_DEFUN([AG_GST_CHECK_DOWHILE_MACROS],[
+
+dnl *** check for working do while(0) macros ***
+AC_CACHE_CHECK([for working do while(0) macros], _cv_g_support_dowhile_macros, [
+ AC_TRY_COMPILE([],[
+ #define STMT_START do
+ #define STMT_END while(0)
+ #define STMT_TEST STMT_START { i = 0; } STMT_END
+ int main(void) { int i = 1; STMT_TEST; return i; }],
+ [_cv_g_support_dowhile_macros=yes],
+ [_cv_g_support_dowhile_macros=no],
+ [_cv_g_support_dowhile_macros=yes])
+])
+if test x$_cv_g_support_dowhile_macros = xyes; then
+ AC_DEFINE(HAVE_DOWHILE_MACROS, 1, [define for working do while(0) macros])
+fi
+])
--- /dev/null
+dnl handle various error-related things
+
+dnl Thomas Vander Stichele <thomas@apestaart.org>
+dnl Tim-Philipp Müller <tim centricular net>
+
+dnl Last modification: 2008-02-18
+
+dnl AG_GST_SET_ERROR_CFLAGS([ADD-WERROR], [MORE_FLAGS])
+dnl AG_GST_SET_ERROR_CXXFLAGS([ADD-WERROR], [MORE_FLAGS])
+dnl AG_GST_SET_LEVEL_DEFAULT([IS-GIT-VERSION])
+
+
+dnl Sets WARNING_CFLAGS and ERROR_CFLAGS to something the compiler
+dnl will accept and AC_SUBST them so they are available in Makefile
+dnl
+dnl WARNING_CFLAGS will contain flags to make the compiler emit more
+dnl warnings.
+dnl ERROR_CFLAGS will contain flags to make those warnings fatal,
+dnl unless ADD-WERROR is set to "no"
+dnl
+dnl If MORE_FLAGS is set, tries to add each of the given flags
+dnl to WARNING_CFLAGS if the compiler supports them. Each flag is
+dnl tested separately.
+dnl
+dnl These flags can be overridden at make time:
+dnl make ERROR_CFLAGS=
+AC_DEFUN([AG_GST_SET_ERROR_CFLAGS],
+[
+ AC_REQUIRE([AC_PROG_CC])
+ AC_REQUIRE([AS_COMPILER_FLAG])
+
+ WARNING_CFLAGS=""
+ ERROR_CFLAGS=""
+
+ dnl if we support -Wall, set it unconditionally
+ AS_COMPILER_FLAG(-Wall,
+ WARNING_CFLAGS="$WARNING_CFLAGS -Wall")
+
+ dnl Warn if declarations after statements are used (C99 extension)
+ AS_COMPILER_FLAG(-Wdeclaration-after-statement,
+ WARNING_CFLAGS="$WARNING_CFLAGS -Wdeclaration-after-statement")
+
+ dnl Warn if variable length arrays are used (C99 extension)
+ AS_COMPILER_FLAG(-Wvla,
+ WARNING_CFLAGS="$WARNING_CFLAGS -Wvla")
+
+ dnl Warn for invalid pointer arithmetic
+ AS_COMPILER_FLAG(-Wpointer-arith,
+ WARNING_CFLAGS="$WARNING_CFLAGS -Wpointer-arith")
+
+ dnl if asked for, add -Werror if supported
+ if test "x$1" != "xno"
+ then
+ AS_COMPILER_FLAG(-Werror, ERROR_CFLAGS="$ERROR_CFLAGS -Werror")
+
+ dnl if -Werror isn't suported, try -errwarn=%all (Sun Forte case)
+ if test "x$ERROR_CFLAGS" = "x"
+ then
+ AS_COMPILER_FLAG([-errwarn=%all], [
+ ERROR_CFLAGS="-errwarn=%all"
+ dnl try -errwarn=%all,no%E_EMPTY_DECLARATION,
+ dnl no%E_STATEMENT_NOT_REACHED,no%E_ARGUEMENT_MISMATCH,
+ dnl no%E_MACRO_REDEFINED (Sun Forte case)
+ dnl For Forte we need disable "empty declaration" warning produced by un-needed semicolon
+ dnl "statement not reached" disabled because there is g_assert_not_reached () in some places
+ dnl "macro redefined" because of gst/gettext.h
+ dnl FIXME: is it really supposed to be 'ARGUEMENT' and not 'ARGUMENT'?
+ for f in 'no%E_EMPTY_DECLARATION' \
+ 'no%E_STATEMENT_NOT_REACHED' \
+ 'no%E_ARGUEMENT_MISMATCH' \
+ 'no%E_MACRO_REDEFINED' \
+ 'no%E_LOOP_NOT_ENTERED_AT_TOP'
+ do
+ AS_COMPILER_FLAG([-errwarn=%all,$f], [
+ ERROR_CFLAGS="$ERROR_CFLAGS,$f"
+ ])
+ done
+ ])
+ else
+ dnl Add -fno-strict-aliasing for GLib versions before 2.19.8
+ dnl as before G_LOCK and friends caused strict aliasing compiler
+ dnl warnings.
+ PKG_CHECK_EXISTS([glib-2.0 < 2.19.8], [
+ AS_COMPILER_FLAG(-fno-strict-aliasing,
+ ERROR_CFLAGS="$ERROR_CFLAGS -fno-strict-aliasing")
+ ])
+ fi
+ fi
+
+ if test "x$2" != "x"
+ then
+ UNSUPPORTED=""
+ list="$2"
+ for each in $list
+ do
+ AS_COMPILER_FLAG($each,
+ WARNING_CFLAGS="$WARNING_CFLAGS $each",
+ UNSUPPORTED="$UNSUPPORTED $each")
+ done
+ if test "X$UNSUPPORTED" != X ; then
+ AC_MSG_NOTICE([unsupported compiler flags: $UNSUPPORTED])
+ fi
+ fi
+
+ AC_SUBST(WARNING_CFLAGS)
+ AC_SUBST(ERROR_CFLAGS)
+ AC_MSG_NOTICE([set WARNING_CFLAGS to $WARNING_CFLAGS])
+ AC_MSG_NOTICE([set ERROR_CFLAGS to $ERROR_CFLAGS])
+])
+
+dnl Sets WARNING_CXXFLAGS and ERROR_CXXFLAGS to something the compiler
+dnl will accept and AC_SUBST them so they are available in Makefile
+dnl
+dnl WARNING_CXXFLAGS will contain flags to make the compiler emit more
+dnl warnings.
+dnl ERROR_CXXFLAGS will contain flags to make those warnings fatal,
+dnl unless ADD-WERROR is set to "no"
+dnl
+dnl If MORE_FLAGS is set, tries to add each of the given flags
+dnl to WARNING_CFLAGS if the compiler supports them. Each flag is
+dnl tested separately.
+dnl
+dnl These flags can be overridden at make time:
+dnl make ERROR_CXXFLAGS=
+AC_DEFUN([AG_GST_SET_ERROR_CXXFLAGS],
+[
+ AC_REQUIRE([AC_PROG_CXX])
+ AC_REQUIRE([AS_CXX_COMPILER_FLAG])
+
+ ERROR_CXXFLAGS=""
+ WARNING_CXXFLAGS=""
+
+ dnl if we support -Wall, set it unconditionally
+ AS_CXX_COMPILER_FLAG(-Wall, WARNING_CXXFLAGS="$WARNING_CXXFLAGS -Wall")
+
+ dnl if asked for, add -Werror if supported
+ if test "x$1" != "xno"
+ then
+ AS_CXX_COMPILER_FLAG(-Werror, ERROR_CXXFLAGS="$ERROR_CXXFLAGS -Werror")
+
+ if test "x$ERROR_CXXFLAGS" != "x"
+ then
+ dnl add exceptions
+ AS_CXX_COMPILER_FLAG([-Wno-non-virtual-dtor], ERROR_CXXFLAGS="$ERROR_CXXFLAGS -Wno-non-virtual-dtor")
+
+ dnl Add -fno-strict-aliasing for GLib versions before 2.19.8
+ dnl as before G_LOCK and friends caused strict aliasing compiler
+ dnl warnings.
+ PKG_CHECK_EXISTS([glib-2.0 < 2.19.8], [
+ AS_CXX_COMPILER_FLAG([-fno-strict-aliasing],
+ ERROR_CXXFLAGS="$ERROR_CXXFLAGS -fno-strict-aliasing")
+ ])
+ else
+ dnl if -Werror isn't suported, try -errwarn=%all
+ AS_CXX_COMPILER_FLAG([-errwarn=%all], ERROR_CXXFLAGS="$ERROR_CXXFLAGS -errwarn=%all")
+ if test "x$ERROR_CXXFLAGS" != "x"; then
+ dnl try -errwarn=%all,no%E_EMPTY_DECLARATION,
+ dnl no%E_STATEMENT_NOT_REACHED,no%E_ARGUEMENT_MISMATCH,
+ dnl no%E_MACRO_REDEFINED (Sun Forte case)
+ dnl For Forte we need disable "empty declaration" warning produced by un-needed semicolon
+ dnl "statement not reached" disabled because there is g_assert_not_reached () in some places
+ dnl "macro redefined" because of gst/gettext.h
+ dnl FIXME: is it really supposed to be 'ARGUEMENT' and not 'ARGUMENT'?
+ dnl FIXME: do any of these work with the c++ compiler? if not, why
+ dnl do we check at all?
+ for f in 'no%E_EMPTY_DECLARATION' \
+ 'no%E_STATEMENT_NOT_REACHED' \
+ 'no%E_ARGUEMENT_MISMATCH' \
+ 'no%E_MACRO_REDEFINED' \
+ 'no%E_LOOP_NOT_ENTERED_AT_TOP'
+ do
+ AS_CXX_COMPILER_FLAG([-errwarn=%all,$f], [ERROR_CXXFLAGS="$ERROR_CXXFLAGS,$f"])
+ done
+ fi
+ fi
+ fi
+
+ if test "x$2" != "x"
+ then
+ UNSUPPORTED=""
+ list="$2"
+ for each in $list
+ do
+ AS_CXX_COMPILER_FLAG($each,
+ WARNING_CXXFLAGS="$WARNING_CXXFLAGS $each",
+ UNSUPPORTED="$UNSUPPORTED $each")
+ done
+ if test "X$UNSUPPORTED" != X ; then
+ AC_MSG_NOTICE([unsupported compiler flags: $UNSUPPORTED])
+ fi
+ fi
+
+ AC_SUBST(WARNING_CXXFLAGS)
+ AC_SUBST(ERROR_CXXFLAGS)
+ AC_MSG_NOTICE([set WARNING_CXXFLAGS to $WARNING_CXXFLAGS])
+ AC_MSG_NOTICE([set ERROR_CXXFLAGS to $ERROR_CXXFLAGS])
+])
+
+dnl Sets WARNING_OBJCFLAGS and ERROR_OBJCFLAGS to something the compiler
+dnl will accept and AC_SUBST them so they are available in Makefile
+dnl
+dnl WARNING_OBJCFLAGS will contain flags to make the compiler emit more
+dnl warnings.
+dnl ERROR_OBJCFLAGS will contain flags to make those warnings fatal,
+dnl unless ADD-WERROR is set to "no"
+dnl
+dnl If MORE_FLAGS is set, tries to add each of the given flags
+dnl to WARNING_CFLAGS if the compiler supports them. Each flag is
+dnl tested separately.
+dnl
+dnl These flags can be overridden at make time:
+dnl make ERROR_OBJCFLAGS=
+AC_DEFUN([AG_GST_SET_ERROR_OBJCFLAGS],
+[
+ AC_REQUIRE([AC_PROG_OBJC])
+ AC_REQUIRE([AS_OBJC_COMPILER_FLAG])
+
+ ERROR_OBJCFLAGS=""
+ WARNING_OBJCFLAGS=""
+
+ dnl if we support -Wall, set it unconditionally
+ AS_OBJC_COMPILER_FLAG(-Wall, WARNING_OBJCFLAGS="$WARNING_OBJCFLAGS -Wall")
+
+ dnl if asked for, add -Werror if supported
+ if test "x$1" != "xno"
+ then
+ AS_OBJC_COMPILER_FLAG(-Werror, ERROR_OBJCFLAGS="$ERROR_OBJCFLAGS -Werror")
+
+ if test "x$ERROR_OBJCFLAGS" != "x"
+ then
+ dnl Add -fno-strict-aliasing for GLib versions before 2.19.8
+ dnl as before G_LOCK and friends caused strict aliasing compiler
+ dnl warnings.
+ PKG_CHECK_EXISTS([glib-2.0 < 2.19.8], [
+ AS_OBJC_COMPILER_FLAG([-fno-strict-aliasing],
+ ERROR_OBJCFLAGS="$ERROR_OBJCFLAGS -fno-strict-aliasing")
+ ])
+ else
+ dnl if -Werror isn't suported, try -errwarn=%all
+ AS_OBJC_COMPILER_FLAG([-errwarn=%all], ERROR_OBJCFLAGS="$ERROR_OBJCFLAGS -errwarn=%all")
+ if test "x$ERROR_OBJCFLAGS" != "x"; then
+ dnl try -errwarn=%all,no%E_EMPTY_DECLARATION,
+ dnl no%E_STATEMENT_NOT_REACHED,no%E_ARGUEMENT_MISMATCH,
+ dnl no%E_MACRO_REDEFINED (Sun Forte case)
+ dnl For Forte we need disable "empty declaration" warning produced by un-needed semicolon
+ dnl "statement not reached" disabled because there is g_assert_not_reached () in some places
+ dnl "macro redefined" because of gst/gettext.h
+ dnl FIXME: is it really supposed to be 'ARGUEMENT' and not 'ARGUMENT'?
+ dnl FIXME: do any of these work with the c++ compiler? if not, why
+ dnl do we check at all?
+ for f in 'no%E_EMPTY_DECLARATION' \
+ 'no%E_STATEMENT_NOT_REACHED' \
+ 'no%E_ARGUEMENT_MISMATCH' \
+ 'no%E_MACRO_REDEFINED' \
+ 'no%E_LOOP_NOT_ENTERED_AT_TOP'
+ do
+ AS_OBJC_COMPILER_FLAG([-errwarn=%all,$f], [ERROR_OBJCFLAGS="$ERROR_OBJCFLAGS,$f"])
+ done
+ fi
+ fi
+ fi
+
+ if test "x$2" != "x"
+ then
+ UNSUPPORTED=""
+ list="$2"
+ for each in $list
+ do
+ AS_OBJC_COMPILER_FLAG($each,
+ WARNING_OBJCFLAGS="$WARNING_OBJCFLAGS $each",
+ UNSUPPORTED="$UNSUPPORTED $each")
+ done
+ if test "X$UNSUPPORTED" != X ; then
+ AC_MSG_NOTICE([unsupported compiler flags: $UNSUPPORTED])
+ fi
+ fi
+
+ AC_SUBST(WARNING_OBJCFLAGS)
+ AC_SUBST(ERROR_OBJCFLAGS)
+ AC_MSG_NOTICE([set WARNING_OBJCFLAGS to $WARNING_OBJCFLAGS])
+ AC_MSG_NOTICE([set ERROR_OBJCFLAGS to $ERROR_OBJCFLAGS])
+])
+
+dnl Sets the default error level for debugging messages
+AC_DEFUN([AG_GST_SET_LEVEL_DEFAULT],
+[
+ dnl define correct errorlevel for debugging messages. We want to have
+ dnl GST_ERROR messages printed when running cvs builds
+ if test "x[$1]" = "xyes"; then
+ GST_LEVEL_DEFAULT=GST_LEVEL_ERROR
+ else
+ GST_LEVEL_DEFAULT=GST_LEVEL_NONE
+ fi
+ AC_DEFINE_UNQUOTED(GST_LEVEL_DEFAULT, $GST_LEVEL_DEFAULT,
+ [Default errorlevel to use])
+ dnl AC_SUBST so we can use it for win32/common/config.h
+ AC_SUBST(GST_LEVEL_DEFAULT)
+])
--- /dev/null
+dnl Perform a check for a feature for GStreamer
+dnl Richard Boulton <richard-alsa@tartarus.org>
+dnl Thomas Vander Stichele <thomas@apestaart.org> added useful stuff
+dnl Last modification: 25/06/2001
+dnl
+dnl AG_GST_CHECK_FEATURE(FEATURE-NAME, FEATURE-DESCRIPTION,
+dnl DEPENDENT-PLUGINS, TEST-FOR-FEATURE,
+dnl DISABLE-BY-DEFAULT, ACTION-IF-USE, ACTION-IF-NOTUSE)
+dnl
+dnl This macro adds a command line argument to allow the user to enable
+dnl or disable a feature, and if the feature is enabled, performs a supplied
+dnl test to check if the feature is available.
+dnl
+dnl The test should define HAVE_<FEATURE-NAME> to "yes" or "no" depending
+dnl on whether the feature is available.
+dnl
+dnl The macro will set USE_<FEATURE-NAME> to "yes" or "no" depending on
+dnl whether the feature is to be used.
+dnl Thomas changed this, so that when USE_<FEATURE-NAME> was already set
+dnl to no, then it stays that way.
+dnl
+dnl The macro will call AM_CONDITIONAL(USE_<FEATURE-NAME>, ...) to allow
+dnl the feature to control what is built in Makefile.ams. If you want
+dnl additional actions resulting from the test, you can add them with the
+dnl ACTION-IF-USE and ACTION-IF-NOTUSE parameters.
+dnl
+dnl FEATURE-NAME is the name of the feature, and should be in
+dnl purely upper case characters.
+dnl FEATURE-DESCRIPTION is used to describe the feature in help text for
+dnl the command line argument.
+dnl DEPENDENT-PLUGINS lists any plug-ins which depend on this feature.
+dnl TEST-FOR-FEATURE is a test which sets HAVE_<FEATURE-NAME> to "yes"
+dnl or "no" depending on whether the feature is
+dnl available.
+dnl DISABLE-BY-DEFAULT if "disabled", the feature is disabled by default,
+dnl if any other value, the feature is enabled by default.
+dnl ACTION-IF-USE any extra actions to perform if the feature is to be
+dnl used.
+dnl ACTION-IF-NOTUSE any extra actions to perform if the feature is not to
+dnl be used.
+dnl
+dnl
+dnl thomas :
+dnl we also added a history.
+dnl GST_PLUGINS_YES will contain all plugins to be built
+dnl that were checked through AG_GST_CHECK_FEATURE
+dnl GST_PLUGINS_NO will contain those that won't be built
+
+AC_DEFUN([AG_GST_CHECK_FEATURE],
+[echo
+AC_MSG_NOTICE(*** checking feature: [$2] ***)
+if test "x[$3]" != "x"
+then
+ AC_MSG_NOTICE(*** for plug-ins: [$3] ***)
+fi
+dnl
+builtin(define, [gst_endisable], ifelse($5, [disabled], [enable], [disable]))dnl
+dnl if it is set to NO, then don't even consider it for building
+NOUSE=
+if test "x$USE_[$1]" = "xno"; then
+ NOUSE="yes"
+fi
+AC_ARG_ENABLE(translit([$1], A-Z, a-z),
+ [ ]builtin(format, --%-26s gst_endisable %s, gst_endisable-translit([$1], A-Z, a-z), [$2]ifelse([$3],,,: [$3])),
+ [ case "${enableval}" in
+ yes) USE_[$1]=yes;;
+ no) USE_[$1]=no;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --enable-translit([$1], A-Z, a-z)) ;;
+ esac],
+ [ USE_$1=]ifelse($5, [disabled], [no], [yes])) dnl DEFAULT
+
+dnl *** set it back to no if it was preset to no
+if test "x$NOUSE" = "xyes"; then
+ USE_[$1]="no"
+ AC_MSG_WARN(*** $3 pre-configured not to be built)
+fi
+NOUSE=
+
+dnl *** Check if it is ported or not
+if echo " [$GST_PLUGINS_NONPORTED] " | tr , ' ' | grep -i " [$1] " > /dev/null; then
+ USE_[$1]="no"
+ AC_MSG_WARN(*** $3 not ported)
+fi
+
+dnl *** If it's enabled
+
+if test x$USE_[$1] = xyes; then
+ dnl save compile variables before the test
+
+ gst_check_save_LIBS=$LIBS
+ gst_check_save_LDFLAGS=$LDFLAGS
+ gst_check_save_CFLAGS=$CFLAGS
+ gst_check_save_CPPFLAGS=$CPPFLAGS
+ gst_check_save_CXXFLAGS=$CXXFLAGS
+
+ HAVE_[$1]=no
+ dnl TEST_FOR_FEATURE
+ $4
+
+ LIBS=$gst_check_save_LIBS
+ LDFLAGS=$gst_check_save_LDFLAGS
+ CFLAGS=$gst_check_save_CFLAGS
+ CPPFLAGS=$gst_check_save_CPPFLAGS
+ CXXFLAGS=$gst_check_save_CXXFLAGS
+
+ dnl If it isn't found, unset USE_[$1]
+ if test x$HAVE_[$1] = xno; then
+ USE_[$1]=no
+ else
+ ifelse([$3], , :, [AC_MSG_NOTICE(*** These plugins will be built: [$3])])
+ fi
+fi
+dnl *** Warn if it's disabled or not found
+if test x$USE_[$1] = xyes; then
+ ifelse([$6], , :, [$6])
+ if test "x$3" != "x"; then
+ GST_PLUGINS_YES="\t[$3]\n$GST_PLUGINS_YES"
+ fi
+ AC_DEFINE(HAVE_[$1], , [Define to enable $2]ifelse($3,,, [ (used by $3)]).)
+else
+ ifelse([$3], , :, [AC_MSG_NOTICE(*** These plugins will not be built: [$3])])
+ if test "x$3" != "x"; then
+ GST_PLUGINS_NO="\t[$3]\n$GST_PLUGINS_NO"
+ fi
+ ifelse([$7], , :, [$7])
+fi
+dnl *** Define the conditional as appropriate
+AM_CONDITIONAL(USE_[$1], test x$USE_[$1] = xyes)
+])
+
+dnl Use AC_CHECK_LIB and AC_CHECK_HEADER to do both tests at once
+dnl sets HAVE_module if we have it
+dnl Richard Boulton <richard-alsa@tartarus.org>
+dnl Last modification: 26/06/2001
+dnl AG_GST_CHECK_LIBHEADER(FEATURE-NAME, LIB NAME, LIB FUNCTION, EXTRA LD FLAGS,
+dnl HEADER NAME, ACTION-IF-FOUND, ACTION-IF-NOT-FOUND)
+dnl
+dnl This check was written for GStreamer: it should be renamed and checked
+dnl for portability if you decide to use it elsewhere.
+dnl
+AC_DEFUN([AG_GST_CHECK_LIBHEADER],
+[
+ AC_CHECK_LIB([$2], [$3], HAVE_[$1]=yes, HAVE_[$1]=no,[$4])
+ if test "x$HAVE_[$1]" = "xyes"; then
+ AC_CHECK_HEADER([$5], :, HAVE_[$1]=no)
+ if test "x$HAVE_[$1]" = "xyes"; then
+ dnl execute what needs to be
+ ifelse([$6], , :, [$6])
+ else
+ ifelse([$7], , :, [$7])
+ fi
+ else
+ ifelse([$7], , :, [$7])
+ fi
+ AC_SUBST(HAVE_[$1])
+]
+)
+
+dnl 2004-02-14 Thomas - changed to get set properly and use proper output
+dnl 2003-06-27 Benjamin Otte - changed to make this work with gstconfig.h
+dnl
+dnl Add a subsystem --disable flag and all the necessary symbols and substitions
+dnl
+dnl AG_GST_CHECK_SUBSYSTEM_DISABLE(SYSNAME, [subsystem name])
+dnl
+AC_DEFUN([AG_GST_CHECK_SUBSYSTEM_DISABLE],
+[
+ dnl this define will replace each literal subsys_def occurrence with
+ dnl the lowercase hyphen-separated subsystem
+ dnl e.g. if $1 is GST_DEBUG then subsys_def will be a macro with gst-debug
+ define([subsys_def],translit([$1], _A-Z, -a-z))
+
+ AC_ARG_ENABLE(subsys_def,
+ AC_HELP_STRING(--disable-subsys_def, [disable $2]),
+ [
+ case "${enableval}" in
+ yes) GST_DISABLE_[$1]=no ;;
+ no) GST_DISABLE_[$1]=yes ;;
+ *) AC_MSG_ERROR([bad value ${enableval} for --enable-subsys_def]) ;;
+ esac
+ ],
+ [GST_DISABLE_[$1]=no]) dnl Default value
+
+ if test x$GST_DISABLE_[$1] = xyes; then
+ AC_MSG_NOTICE([disabled subsystem [$2]])
+ GST_DISABLE_[$1]_DEFINE="#define GST_DISABLE_$1 1"
+ else
+ GST_DISABLE_[$1]_DEFINE="/* #undef GST_DISABLE_$1 */"
+ fi
+ AC_SUBST(GST_DISABLE_[$1]_DEFINE)
+ undefine([subsys_def])
+])
+
+
+dnl Parse gstconfig.h for feature and defines add the symbols and substitions
+dnl
+dnl AG_GST_PARSE_SUBSYSTEM_DISABLE(GST_CONFIGPATH, FEATURE)
+dnl
+AC_DEFUN([AG_GST_PARSE_SUBSYSTEM_DISABLE],
+[
+ grep >/dev/null "#undef GST_DISABLE_$2" $1
+ if test $? = 0; then
+ GST_DISABLE_[$2]=0
+ else
+ GST_DISABLE_[$2]=1
+ fi
+ AC_SUBST(GST_DISABLE_[$2])
+])
+
+dnl Parse gstconfig.h and defines add the symbols and substitions
+dnl
+dnl GST_CONFIGPATH=`$PKG_CONFIG --variable=includedir gstreamer-1.0`"/gst/gstconfig.h"
+dnl AG_GST_PARSE_SUBSYSTEM_DISABLES(GST_CONFIGPATH)
+dnl
+AC_DEFUN([AG_GST_PARSE_SUBSYSTEM_DISABLES],
+[
+ AG_GST_PARSE_SUBSYSTEM_DISABLE($1,GST_DEBUG)
+ AG_GST_PARSE_SUBSYSTEM_DISABLE($1,LOADSAVE)
+ AG_GST_PARSE_SUBSYSTEM_DISABLE($1,PARSE)
+ AG_GST_PARSE_SUBSYSTEM_DISABLE($1,TRACE)
+ AG_GST_PARSE_SUBSYSTEM_DISABLE($1,ALLOC_TRACE)
+ AG_GST_PARSE_SUBSYSTEM_DISABLE($1,REGISTRY)
+ AG_GST_PARSE_SUBSYSTEM_DISABLE($1,PLUGIN)
+ AG_GST_PARSE_SUBSYSTEM_DISABLE($1,XML)
+])
+
+dnl AG_GST_CHECK_GST_DEBUG_DISABLED(ACTION-IF-DISABLED, ACTION-IF-NOT-DISABLED)
+dnl
+dnl Checks if the GStreamer debugging system is disabled in the core version
+dnl we are compiling against (by checking gstconfig.h)
+dnl
+AC_DEFUN([AG_GST_CHECK_GST_DEBUG_DISABLED],
+[
+ AC_REQUIRE([AG_GST_CHECK_GST])
+
+ AC_MSG_CHECKING([whether the GStreamer debugging system is enabled])
+ AC_LANG_PUSH([C])
+ save_CFLAGS="$CFLAGS"
+ CFLAGS="$GST_CFLAGS $CFLAGS"
+ AC_COMPILE_IFELSE([
+ AC_LANG_SOURCE([[
+ #include <gst/gstconfig.h>
+ #ifdef GST_DISABLE_GST_DEBUG
+ #error "debugging disabled, make compiler fail"
+ #endif]])], [ debug_system_enabled=yes], [debug_system_enabled=no])
+ CFLAGS="$save_CFLAGS"
+ AC_LANG_POP([C])
+
+ AC_MSG_RESULT([$debug_system_enabled])
+
+ if test "x$debug_system_enabled" = "xyes" ; then
+ $2
+ true
+ else
+ $1
+ true
+ fi
+])
+
+dnl relies on GST_PLUGINS_ALL, GST_PLUGINS_SELECTED, GST_PLUGINS_YES,
+dnl GST_PLUGINS_NO, and BUILD_EXTERNAL
+AC_DEFUN([AG_GST_OUTPUT_PLUGINS], [
+
+printf "configure: *** Plug-ins without external dependencies that will be built:\n"
+( for i in $GST_PLUGINS_SELECTED; do printf '\t'$i'\n'; done ) | sort
+printf "\n"
+
+printf "configure: *** Plug-ins without external dependencies that will NOT be built:\n"
+( for i in $GST_PLUGINS_ALL; do
+ case " $GST_PLUGINS_SELECTED " in
+ *\ $i\ *)
+ ;;
+ *)
+ printf '\t'$i'\n'
+ ;;
+ esac
+ done ) | sort
+printf "\n"
+
+printf "configure: *** Plug-ins that have NOT been ported:\n"
+( for i in $GST_PLUGINS_NONPORTED; do
+ printf '\t'$i'\n'
+ done ) | sort
+printf "\n"
+
+if test "x$BUILD_EXTERNAL" = "xno"; then
+ printf "configure: *** No plug-ins with external dependencies will be built\n"
+else
+ printf "configure: *** Plug-ins with dependencies that will be built:"
+ printf "$GST_PLUGINS_YES\n" | sort
+ printf "\n"
+ printf "configure: *** Plug-ins with dependencies that will NOT be built:"
+ printf "$GST_PLUGINS_NO\n" | sort
+ printf "\n"
+fi
+])
+
--- /dev/null
+dnl
+dnl Check for compiler mechanism to show functions in debugging
+dnl copied from an Ali patch floating on the internet
+dnl
+AC_DEFUN([AG_GST_CHECK_FUNCTION],[
+ dnl #1: __PRETTY_FUNCTION__
+ AC_MSG_CHECKING(whether $CC implements __PRETTY_FUNCTION__)
+ AC_CACHE_VAL(gst_cv_have_pretty_function,[
+ AC_TRY_LINK([#include <stdio.h>],
+ [printf("%s", __PRETTY_FUNCTION__);],
+ gst_cv_have_pretty_function=yes,
+ gst_cv_have_pretty_function=no)
+ ])
+ AC_MSG_RESULT($gst_cv_have_pretty_function)
+ if test "$gst_cv_have_pretty_function" = yes; then
+ AC_DEFINE(HAVE_PRETTY_FUNCTION, 1,
+ [defined if the compiler implements __PRETTY_FUNCTION__])
+ fi
+
+dnl #2: __FUNCTION__
+ AC_MSG_CHECKING(whether $CC implements __FUNCTION__)
+ AC_CACHE_VAL(gst_cv_have_function,[
+ AC_TRY_LINK([#include <stdio.h>],
+ [printf("%s", __FUNCTION__);],
+ gst_cv_have_function=yes,
+ gst_cv_have_function=no)
+ ])
+ AC_MSG_RESULT($gst_cv_have_function)
+ if test "$gst_cv_have_function" = yes; then
+ AC_DEFINE(HAVE_FUNCTION, 1,
+ [defined if the compiler implements __FUNCTION__])
+ fi
+
+dnl #3: __func__
+ AC_MSG_CHECKING(whether $CC implements __func__)
+ AC_CACHE_VAL(gst_cv_have_func,[
+ AC_TRY_LINK([#include <stdio.h>],
+ [printf("%s", __func__);],
+ gst_cv_have_func=yes,
+ gst_cv_have_func=no)
+ ])
+ AC_MSG_RESULT($gst_cv_have_func)
+ if test "$gst_cv_have_func" = yes; then
+ AC_DEFINE(HAVE_FUNC, 1,
+ [defined if the compiler implements __func__])
+ fi
+
+dnl now define FUNCTION to whatever works, and fallback to ""
+ if test "$gst_cv_have_pretty_function" = yes; then
+ function=__PRETTY_FUNCTION__
+ else
+ if test "$gst_cv_have_function" = yes; then
+ function=__FUNCTION__
+ else
+ if test "$gst_cv_have_func" = yes; then
+ function=__func__
+ else
+ function=\"\"
+ fi
+ fi
+ fi
+ AC_DEFINE_UNQUOTED(GST_FUNCTION, $function, [macro to use to show function name])
+])
--- /dev/null
+dnl gettext setup
+
+dnl AG_GST_GETTEXT([gettext-package])
+dnl defines GETTEXT_PACKAGE and LOCALEDIR
+
+AC_DEFUN([AG_GST_GETTEXT],
+[
+ if test "$USE_NLS" = "yes"; then
+ GETTEXT_PACKAGE=[$1]
+ else
+ GETTEXT_PACKAGE=[NULL]
+ fi
+ AC_SUBST(GETTEXT_PACKAGE)
+ AC_DEFINE_UNQUOTED([GETTEXT_PACKAGE], "$GETTEXT_PACKAGE",
+ [gettext package name])
+
+ dnl make sure po/Makevars is kept in sync with GETTEXT_PACKAGE
+ if test -e "${srcdir}/po/Makevars"; then
+ if ! grep -e "$1" "${srcdir}/po/Makevars"; then
+ AC_MSG_ERROR([DOMAIN in po/Makevars does not match GETTEXT_PACKAGE $1])
+ fi
+ fi
+
+ dnl define LOCALEDIR in config.h
+ AS_AC_EXPAND(LOCALEDIR, $datadir/locale)
+ AC_DEFINE_UNQUOTED([LOCALEDIR], "$LOCALEDIR",
+ [gettext locale dir])
+])
--- /dev/null
+dnl check for a minimum version of GLib
+
+dnl AG_GST_GLIB_CHECK([minimum-version-required])
+
+AC_DEFUN([AG_GST_GLIB_CHECK],
+[
+ AC_REQUIRE([AS_NANO])
+
+ dnl Minimum required version of GLib
+ GLIB_REQ=[$1]
+ if test "x$GLIB_REQ" = "x"
+ then
+ AC_MSG_ERROR([Please specify a required version for GLib 2.0])
+ fi
+ AC_SUBST(GLIB_REQ)
+
+ dnl Check for glib with everything
+ AG_GST_PKG_CHECK_MODULES(GLIB,
+ glib-2.0 >= $GLIB_REQ gobject-2.0 gmodule-no-export-2.0)
+
+ if test "x$HAVE_GLIB" = "xno"; then
+ AC_MSG_ERROR([This package requires GLib >= $GLIB_REQ to compile.])
+ fi
+
+ dnl Add define to tell GLib that threading is always enabled within GStreamer
+ dnl code (optimisation, bypasses checks if the threading system is enabled
+ dnl when using threading primitives)
+ GLIB_EXTRA_CFLAGS="$GLIB_EXTRA_CFLAGS -DG_THREADS_MANDATORY"
+
+ dnl Define G_DISABLE_DEPRECATED for development versions
+ if test "x`expr $PACKAGE_VERSION_MINOR % 2`" = "x1" -a "x`expr $PACKAGE_VERSION_MICRO '<' 90`" = "x1"; then
+ GLIB_EXTRA_CFLAGS="$GLIB_EXTRA_CFLAGS -DG_DISABLE_DEPRECATED"
+ fi
+
+ AC_ARG_ENABLE(gobject-cast-checks,
+ AS_HELP_STRING([--enable-gobject-cast-checks[=@<:@no/auto/yes@:>@]],
+ [Enable GObject cast checks]),[enable_gobject_cast_checks=$enableval],
+ [enable_gobject_cast_checks=auto])
+
+ if test "x$enable_gobject_cast_checks" = "xauto"; then
+ dnl Turn on cast checks only for development versions
+ if test "x`expr $PACKAGE_VERSION_MINOR % 2`" = "x1" -a "x`expr $PACKAGE_VERSION_MICRO '<' 90`" = "x1"; then
+ enable_gobject_cast_checks=yes
+ else
+ enable_gobject_cast_checks=no
+ fi
+ fi
+
+ if test "x$enable_gobject_cast_checks" = "xno"; then
+ GLIB_EXTRA_CFLAGS="$GLIB_EXTRA_CFLAGS -DG_DISABLE_CAST_CHECKS"
+ fi
+
+ AC_ARG_ENABLE(glib-asserts,
+ AS_HELP_STRING([--enable-glib-asserts[=@<:@no/auto/yes@:>@]],
+ [Enable GLib assertion]),[enable_glib_assertions=$enableval],
+ [enable_glib_assertions=auto])
+
+ if test "x$enable_glib_assertions" = "xauto"; then
+ dnl Enable assertions only for development versions
+ if test "x`expr $PACKAGE_VERSION_MINOR % 2`" = "x1" -a "x`expr $PACKAGE_VERSION_MICRO '<' 90`" = "x1"; then
+ enable_glib_assertions=yes
+ else
+ enable_glib_assertions=no
+ fi
+ fi
+
+ if test "x$enable_glib_assertions" = "xno"; then
+ GLIB_EXTRA_CFLAGS="$GLIB_EXTRA_CFLAGS -DG_DISABLE_ASSERT"
+ fi
+
+ dnl Find location of glib utils. People may want to or have to override these,
+ dnl e.g. in a cross-compile situation where PATH is a bit messed up. We need
+ dnl for these tools to work on the host, so can't just use the one from the
+ dnl GLib installation that pkg-config picks up, as that might be for a
+ dnl different target architecture.
+ dnl
+ dnl glib-genmarshal:
+ AC_MSG_CHECKING(for glib-genmarshal)
+ if test "x$GLIB_GENMARSHAL" != "x"; then
+ AC_MSG_RESULT([$GLIB_GENMARSHAL (from environment)])
+ else
+ GLIB_GENMARSHAL=`$PKG_CONFIG --variable=glib_genmarshal glib-2.0`
+ if $GLIB_GENMARSHAL --version 2>/dev/null >/dev/null; then
+ AC_MSG_RESULT([$GLIB_GENMARSHAL (from pkg-config path)])
+ else
+ AC_PATH_PROG(GLIB_GENMARSHAL, [glib-genmarshal], [glib-genmarshal])
+ AC_MSG_RESULT([$GLIB_GENMARSHAL])
+ fi
+ fi
+ if ! $GLIB_GENMARSHAL --version 2>/dev/null >/dev/null; then
+ AC_MSG_WARN([$GLIB_GENMARSHAL does not seem to work!])
+ fi
+ AC_SUBST(GLIB_GENMARSHAL)
+
+ dnl glib-mkenums:
+ AC_MSG_CHECKING(for glib-mkenums)
+ if test "x$GLIB_MKENUMS" != "x"; then
+ AC_MSG_RESULT([$GLIB_MKENUMS (from environment)])
+ else
+ dnl glib-mkenums is written in perl so should always work really
+ GLIB_MKENUMS=`$PKG_CONFIG --variable=glib_mkenums glib-2.0`
+ AC_MSG_RESULT([$GLIB_MKENUMS])
+ fi
+ if ! $GLIB_MKENUMS --version 2>/dev/null >/dev/null; then
+ AC_MSG_WARN([$GLIB_MKENUMS does not seem to work!])
+ fi
+ AC_SUBST(GLIB_MKENUMS)
+
+ AC_SUBST(GLIB_EXTRA_CFLAGS)
+
+ dnl Now check for GIO
+ PKG_CHECK_MODULES(GIO, gio-2.0 >= $GLIB_REQ)
+ if test "x$HAVE_GIO" = "xno"; then
+ AC_MSG_ERROR([This package requires GIO >= $GLIB_REQ to compile.])
+ fi
+
+ GIO_MODULE_DIR="`$PKG_CONFIG --variable=giomoduledir gio-2.0`"
+ AC_DEFINE_UNQUOTED(GIO_MODULE_DIR, "$GIO_MODULE_DIR",
+ [The GIO modules directory.])
+ GIO_LIBDIR="`$PKG_CONFIG --variable=libdir gio-2.0`"
+ AC_DEFINE_UNQUOTED(GIO_LIBDIR, "$GIO_LIBDIR",
+ [The GIO library directory.])
+ AC_SUBST(GIO_CFLAGS)
+ AC_SUBST(GIO_LIBS)
+ AC_SUBST(GIO_LDFLAGS)
+])
--- /dev/null
+dnl call this macro with the minimum required version as an argument
+dnl this macro sets and AC_SUBSTs XML_CFLAGS and XML_LIBS
+dnl it also sets LIBXML_PKG, used for the pkg-config file
+
+AC_DEFUN([AG_GST_LIBXML2_CHECK],
+[
+ dnl Minimum required version of libxml2
+ dnl default to 2.4.9 if not specified
+ LIBXML2_REQ=ifelse([$1],,2.4.9,[$1])
+ AC_SUBST(LIBXML2_REQ)
+
+ dnl check for libxml2
+ PKG_CHECK_MODULES(XML, libxml-2.0 >= $LIBXML2_REQ,
+ HAVE_LIBXML2=yes, [
+ AC_MSG_RESULT(no)
+ HAVE_LIBXML2=no
+ ])
+ if test "x$HAVE_LIBXML2" = "xyes"; then
+ AC_DEFINE(HAVE_LIBXML2, 1, [Define if libxml2 is available])
+ else
+ AC_MSG_ERROR([
+ Need libxml2 and development headers/files to build GStreamer.
+
+ You can do without libxml2 if you pass --disable-loadsave to
+ configure, but that breaks ABI, so don't do that unless you
+ are building for an embedded setup and know what you are doing.
+ ])
+ fi
+ dnl this is for the .pc file
+ LIBXML_PKG=', libxml-2.0'
+ AC_SUBST(LIBXML_PKG)
+ AC_SUBST(XML_LIBS)
+ AC_SUBST(XML_CFLAGS)
+
+ dnl XML_LIBS might pull in -lz without zlib actually being on the system, so
+ dnl try linking with these LIBS and CFLAGS
+ ac_save_CFLAGS=$CFLAGS
+ ac_save_LIBS=$LIBS
+ CFLAGS="$CFLAGS $XML_CFLAGS"
+ LIBS="$LIBS $XML_LIBS"
+ AC_TRY_LINK([
+#include <libxml/tree.h>
+#include <stdio.h>
+],[
+/* function body */
+],
+ AC_MSG_NOTICE([Test xml2 program linked]),
+ AC_MSG_ERROR([Could not link libxml2 test program. Check if you have the necessary dependencies.])
+ )
+ CFLAGS="$ac_save_CFLAGS"
+ LIBS="$ac_save_LIBS"
+])
--- /dev/null
+dnl macros to set GST_PACKAGE_RELEASE_DATETIME
+
+dnl ===========================================================================
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME
+dnl
+dnl Usage:
+dnl
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME()
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([no]...)
+dnl sets the release datetime to the current date
+dnl (no = this is not a release, but git or prerelease)
+dnl
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([YYYY-MM-DD])
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([yes], [YYYY-MM-DD])
+dnl sets the release datetime to the specified date (and time, if given)
+dnl (yes = this is a release, not git or prerelease)
+dnl
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([yes], [DOAP-FILE], [RELEASE-VERSION])
+dnl sets the release date to the release date associated with version
+dnl RELEASE-VERSION in the .doap file DOAP-FILE
+dnl (yes = this is a release, not git or prerelease)
+dnl
+dnl We need to treat pre-releases like git because there won't be an entry
+dnl in the .doap file for pre-releases yet, and we don't want to use the
+dnl date of the last release either.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_SET_PACKAGE_RELEASE_DATETIME],
+[
+ dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME()
+ dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([yes]...)
+ if test "x$1" = "xno" -o "x$1" = "x"; then
+ GST_PACKAGE_RELEASE_DATETIME=`date -u "+%Y-%m-%dT%H:%MZ"`
+ elif test "x$1" = "xyes"; then
+ dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([no], ["YYYY-MM-DD"])
+ dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([no], [DOAP-FILE], [RELEASE-VERSION])
+ if ( echo $1 | grep '^20[1-9][0-9]-[0-1][0-9]-[0-3][0-9]' >/dev/null ) ; then
+ GST_PACKAGE_RELEASE_DATETIME=$1
+ else
+ dnl we assume the .doap file contains the date as YYYY-MM-DD
+ YYYY_MM_DD=`sh "${srcdir}/common/extract-release-date-from-doap-file" $3 $2`;
+ if test "x$YYYY_MM_DD" != "x"; then
+ GST_PACKAGE_RELEASE_DATETIME=$YYYY_MM_DD
+ else
+ AC_MSG_ERROR([SET_PACKAGE_RELEASE_DATETIME: could not extract
+ release date for release version $3 from $2])
+ GST_PACKAGE_RELEASE_DATETIME=""
+ fi
+ fi
+ dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([YYYY-MM-DD])
+ elif ( echo $1 | grep '^20[1-9][0-9]-[0-1][0-9]-[0-3][0-9]' >/dev/null ) ; then
+ GST_PACKAGE_RELEASE_DATETIME=$1
+ else
+ AC_MSG_WARN([SET_PACKAGE_RELEASE_DATETIME: invalid first argument])
+ GST_PACKAGE_RELEASE_DATETIME=""
+ fi
+
+ if test "x$GST_PACKAGE_RELEASE_DATETIME" = "x"; then
+ AC_MSG_WARN([Invalid package release date time: $GST_PACKAGE_RELEASE_DATETIME])
+ else
+ AC_MSG_NOTICE([Setting GST_PACKAGE_RELEASE_DATETIME to $GST_PACKAGE_RELEASE_DATETIME])
+
+ AC_DEFINE_UNQUOTED([GST_PACKAGE_RELEASE_DATETIME],
+ ["$GST_PACKAGE_RELEASE_DATETIME"],
+ [GStreamer package release date/time for plugins as YYYY-MM-DD])
+ fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO
+dnl
+dnl Usage:
+dnl
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([NANO-VERSION], [DOAP-FILE], [RELEASE-VERSION])
+dnl if NANO-VERSION is 0, sets the release date to the release date associated
+dnl with version RELEASE-VERSION in the .doap file DOAP-FILE, otherwise sets
+dnl the release date and time to the current date/time.
+dnl
+dnl We need to treat pre-releases like git because there won't be an entry
+dnl in the .doap file for pre-releases yet, and we don't want to use the
+dnl date of the last release either.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO],
+[
+ if test "x$1" = "x0"; then
+ AG_GST_SET_PACKAGE_RELEASE_DATETIME([yes], [ $2 ], [ $3 ])
+ else
+ AG_GST_SET_PACKAGE_RELEASE_DATETIME([no])
+ fi
+])
--- /dev/null
+AC_DEFUN([AG_GST_BISON_CHECK],
+[
+ dnl FIXME: check if AC_PROG_YACC is suitable here
+ dnl FIXME: make precious
+ AC_PATH_PROG(BISON_PATH, bison, no)
+ if test x$BISON_PATH = xno; then
+ AC_MSG_ERROR(Could not find bison)
+ fi
+
+ dnl check bison version
+ dnl we need version >= 2.4 for the '<>' support
+ dnl in the parser.
+ dnl First lines observed: 'bison (GNU Bison) 2.3' or 'GNU Bison version 1.28'
+ bison_min_version=2.4
+ bison_version=`$BISON_PATH --version | head -n 1 | sed 's/^[[^0-9]]*//' | sed 's/[[^0-9]]*$//' | cut -d' ' -f1`
+ AC_MSG_CHECKING([bison version $bison_version >= $bison_min_version])
+
+ if perl -we "exit ((v$bison_version ge v$bison_min_version) ? 0 : 1)"; then
+ AC_MSG_RESULT([yes])
+ else
+ AC_MSG_ERROR([no])
+ fi
+])
+
+AC_DEFUN([AG_GST_FLEX_CHECK],
+[
+ dnl we require flex for building the parser
+ AC_PATH_PROG(FLEX_PATH, flex, no)
+ if test x$FLEX_PATH = xno; then
+ AC_MSG_ERROR(Could not find flex)
+ fi
+
+ dnl check flex version
+ dnl we need version >= 2.5.31 for the reentrancy support
+ dnl in the parser.
+ flex_min_version=2.5.31
+ flex_version=`$FLEX_PATH --version | head -n 1 | awk '{print $2}'`
+ AC_MSG_CHECKING([flex version $flex_version >= $flex_min_version])
+ if perl -w <<EOF
+ (\$min_version_major, \$min_version_minor, \$min_version_micro ) = "$flex_min_version" =~ /(\d+)\.(\d+)\.(\d+)/;
+ (\$flex_version_major, \$flex_version_minor, \$flex_version_micro ) = "$flex_version" =~ /(\d+)\.(\d+)\.(\d+)/;
+ exit (((\$flex_version_major > \$min_version_major) ||
+ ((\$flex_version_major == \$min_version_major) &&
+ (\$flex_version_minor > \$min_version_minor)) ||
+ ((\$flex_version_major == \$min_version_major) &&
+ (\$flex_version_minor == \$min_version_minor) &&
+ (\$flex_version_micro >= \$min_version_micro)))
+ ? 0 : 1);
+EOF
+ then
+ AC_MSG_RESULT(yes)
+ else
+ AC_MSG_ERROR([no])
+ fi
+])
--- /dev/null
+dnl AG_GST_PLATFORM
+dnl Check for platform specific features and define some variables
+dnl
+dnl GST_EXTRA_MODULE_SUFFIX: contains a platform specific
+dnl extra module suffix additional to G_MODULE_SUFFIX
+dnl
+dnl HAVE_OSX: Defined if compiling for OS X
+dnl
+dnl GST_HAVE_UNSAFE_FORK: Defined if fork is unsafe (Windows)
+dnl
+dnl HAVE_WIN32: Defined if compiling on Win32
+dnl
+
+AC_DEFUN([AG_GST_PLATFORM],
+[
+ AC_REQUIRE([AC_CANONICAL_HOST])
+
+ case $host_os in
+ rhapsody*)
+ AC_DEFINE_UNQUOTED(GST_EXTRA_MODULE_SUFFIX, [".dylib"], [Extra platform specific plugin suffix])
+ ;;
+ darwin*)
+ AC_DEFINE_UNQUOTED(GST_EXTRA_MODULE_SUFFIX, [".dylib"], [Extra platform specific plugin suffix])
+ AC_DEFINE_UNQUOTED(HAVE_OSX, 1, [Defined if compiling for OSX])
+ ;;
+ cygwin*)
+ AC_DEFINE_UNQUOTED(GST_HAVE_UNSAFE_FORK, 1, [Defined when registry scanning through fork is unsafe])
+ ;;
+ mingw* | msvc* | mks*)
+ dnl HAVE_WIN32 currently means "disable POSIXisms".
+ AC_DEFINE_UNQUOTED(HAVE_WIN32, 1, [Defined if compiling for Windows])
+
+ dnl define __MSVCRT_VERSION__ version if not set already by the
+ dnl compiler (ie. mostly for mingw). This is needed for things like
+ dnl __stat64 to be available. If set by the compiler, ensure it's
+ dnl new enough - we need at least WinXP SP2.
+ AC_TRY_COMPILE([ ], [ return __MSVCRT_VERSION__; ], [
+ AC_TRY_COMPILE([ ], [
+ #if __MSVCRT_VERSION__ < 0x0601
+ #error "MSVCRT too old"
+ #endif
+ ], [
+ AC_MSG_NOTICE([MSVCRT version looks ok])
+ ], [
+ AC_MSG_ERROR([MSVCRT version too old, need at least WinXP SP2])
+ ])
+ ], [
+ AC_MSG_NOTICE([Setting MSVCRT version to 0x0601])
+ AC_DEFINE_UNQUOTED(__MSVCRT_VERSION__, 0x0601, [We need at least WinXP SP2 for __stat64])
+ ])
+ ;;
+ *)
+ ;;
+ esac
+])
+
+AC_DEFUN([AG_GST_LIBTOOL_PREPARE],
+[
+ dnl Persuade libtool to also link (-l) a 'pure' (DirectX) static lib,
+ dnl i.e. as opposed to only import lib with dll counterpart.
+ dnl Needs to be tweaked before libtool's checks.
+ case $host_os in
+ cygwin* | mingw*)
+ lt_cv_deplibs_check_method=pass_all
+ ;;
+ esac
+])
\ No newline at end of file
--- /dev/null
+dnl AG_GST_PLUGIN_DOCS([MINIMUM-GTK-DOC-VERSION])
+dnl
+dnl checks for prerequisites for the common/mangle-tmpl.py script
+dnl used when building the plugin documentation
+
+AC_DEFUN([AG_GST_PLUGIN_DOCS],
+[
+ AC_BEFORE([GTK_DOC_CHECK],[$0])dnl check for gtk-doc first
+ AC_REQUIRE([AM_PATH_PYTHON])dnl find python first
+
+ build_plugin_docs=no
+ AC_MSG_CHECKING([whether to build plugin documentation])
+ if test x$enable_gtk_doc = xyes; then
+ if test x$PYTHON != x; then
+ build_plugin_docs=yes
+ AC_MSG_RESULT([yes])
+ else
+ AC_MSG_RESULT([no (python not found)])
+ fi
+ else
+ AC_MSG_RESULT([no (gtk-doc disabled or not available)])
+ fi
+
+ AM_CONDITIONAL(ENABLE_PLUGIN_DOCS, test x$build_plugin_docs = xyes)
+])
--- /dev/null
+dnl AG_GST_SET_PLUGINDIR
+
+dnl AC_DEFINE PLUGINDIR to the full location where plug-ins will be installed
+dnl AC_SUBST plugindir, to be used in Makefile.am's
+
+AC_DEFUN([AG_GST_SET_PLUGINDIR],
+[
+ dnl define location of plugin directory
+ AS_AC_EXPAND(PLUGINDIR, ${libdir}/gstreamer-$GST_API_VERSION)
+ AC_DEFINE_UNQUOTED(PLUGINDIR, "$PLUGINDIR",
+ [directory where plugins are located])
+ AC_MSG_NOTICE([Using $PLUGINDIR as the plugin install location])
+
+ dnl plugin directory configure-time variable for use in Makefile.am
+ plugindir="\$(libdir)/gstreamer-$GST_API_VERSION"
+ AC_SUBST(plugindir)
+])
--- /dev/null
+AC_DEFUN([AG_GST_VALGRIND_CHECK],
+[
+ dnl valgrind inclusion
+ AC_ARG_ENABLE(valgrind,
+ AC_HELP_STRING([--disable-valgrind], [disable run-time valgrind detection]),
+ [
+ case "${enableval}" in
+ yes) USE_VALGRIND="$USE_DEBUG" ;;
+ no) USE_VALGRIND=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --enable-valgrind) ;;
+ esac],
+ [
+ USE_VALGRIND="$USE_DEBUG"
+ ]) dnl Default value
+
+ VALGRIND_REQ="3.0"
+ if test "x$USE_VALGRIND" = xyes; then
+ PKG_CHECK_MODULES(VALGRIND, valgrind >= $VALGRIND_REQ,
+ USE_VALGRIND="yes",
+ [
+ USE_VALGRIND="no"
+ AC_MSG_RESULT([no])
+ ])
+ fi
+
+ if test "x$USE_VALGRIND" = xyes; then
+ AC_DEFINE(HAVE_VALGRIND, 1, [Define if valgrind should be used])
+ AC_MSG_NOTICE(Using extra code paths for valgrind)
+ fi
+ AC_SUBST(VALGRIND_CFLAGS)
+ AC_SUBST(VALGRIND_LIBS)
+
+ AC_PATH_PROG(VALGRIND_PATH, valgrind, no)
+ AM_CONDITIONAL(HAVE_VALGRIND, test ! "x$VALGRIND_PATH" = "xno")
+])
--- /dev/null
+dnl macros for X-related detections
+dnl AC_SUBST's HAVE_X, X_CFLAGS, X_LIBS
+AC_DEFUN([AG_GST_CHECK_X],
+[
+ AC_PATH_XTRA
+ ac_cflags_save="$CFLAGS"
+ ac_cppflags_save="$CPPFLAGS"
+ CFLAGS="$CFLAGS $X_CFLAGS"
+ CPPFLAGS="$CPPFLAGS $X_CFLAGS"
+
+ dnl now try to find the HEADER
+ AC_CHECK_HEADER(X11/Xlib.h, HAVE_X="yes", HAVE_X="no")
+
+ if test "x$HAVE_X" = "xno"
+ then
+ AC_MSG_NOTICE([cannot find X11 development files])
+ else
+ dnl this is much more than we want
+ X_LIBS="$X_LIBS $X_PRE_LIBS $X_EXTRA_LIBS"
+ dnl AC_PATH_XTRA only defines the path needed to find the X libs,
+ dnl it does not add the libs; therefore we add them here
+ X_LIBS="$X_LIBS -lX11"
+ AC_SUBST(X_CFLAGS)
+ AC_SUBST(X_LIBS)
+ fi
+ AC_SUBST(HAVE_X)
+
+ CFLAGS="$ac_cflags_save"
+ CPPFLAGS="$ac_cppflags_save"
+])
+
+dnl *** XVideo ***
+dnl Look for the PIC library first, Debian requires it.
+dnl Check debian-devel archives for gory details.
+dnl 20020110:
+dnl At the moment XFree86 doesn't distribute shared libXv due
+dnl to unstable API. On many platforms you CAN NOT link a shared
+dnl lib to a static non-PIC lib. This is what the xvideo GStreamer
+dnl plug-in wants to do. So Debian distributes a PIC compiled
+dnl version of the static lib for plug-ins to link to when it is
+dnl inappropriate to link the main application to libXv directly.
+dnl FIXME: add check if this platform can support linking to a
+dnl non-PIC libXv, if not then don not use Xv.
+dnl FIXME: perhaps warn user if they have a shared libXv since
+dnl this is an error until XFree86 starts shipping one
+AC_DEFUN([AG_GST_CHECK_XV],
+[
+ if test x$HAVE_X = xyes; then
+ AC_CHECK_LIB(Xv_pic, XvQueryExtension,
+ HAVE_XVIDEO="yes", HAVE_XVIDEO="no",
+ $X_LIBS -lXext)
+
+ if test x$HAVE_XVIDEO = xyes; then
+ XVIDEO_LIBS="-lXv_pic -lXext"
+ AC_SUBST(XVIDEO_LIBS)
+ else
+ dnl try again using something else if we didn't find it first
+ if test x$HAVE_XVIDEO = xno; then
+ AC_CHECK_LIB(Xv, XvQueryExtension,
+ HAVE_XVIDEO="yes", HAVE_XVIDEO="no",
+ $X_LIBS -lXext)
+
+ if test x$HAVE_XVIDEO = xyes; then
+ XVIDEO_LIBS="-lXv -lXext"
+ AC_SUBST(XVIDEO_LIBS)
+ fi
+ fi
+ fi
+ fi
+])
--- /dev/null
+dnl AG_GST_INIT
+dnl sets up use of GStreamer configure.ac macros
+dnl all GStreamer autoconf macros are prefixed
+dnl with AG_GST_ for public macros
+dnl with _AG_GST_ for private macros
+dnl
+dnl We call AC_CANONICAL_TARGET and AC_CANONICAL_HOST so that
+dnl it is valid before AC_ARG_PROGRAM is called
+
+AC_DEFUN([AG_GST_INIT],
+[
+ m4_pattern_forbid(^_?AG_GST_)
+ AC_REQUIRE([AC_CANONICAL_HOST]) dnl we use host_ variables
+ AC_REQUIRE([AC_CANONICAL_TARGET]) dnl we use target_ variables
+])
+
+dnl AG_GST_PKG_CONFIG_PATH
+dnl
+dnl sets up a GST_PKG_CONFIG_PATH variable for use in Makefile.am
+dnl which contains the path of the in-tree pkgconfig directory first
+dnl and then any paths specified in PKG_CONFIG_PATH.
+dnl
+dnl We do this mostly so we don't have to use unportable shell constructs
+dnl such as ${PKG_CONFIG_PATH:+:$PKG_CONFIG_PATH} in Makefile.am to handle
+dnl the case where the environment variable is not set, but also in order
+dnl to avoid a trailing ':' in the PKG_CONFIG_PATH which apparently causes
+dnl problems with pkg-config on windows with msys/mingw.
+AC_DEFUN([AG_GST_PKG_CONFIG_PATH],
+[
+ GST_PKG_CONFIG_PATH="\$(top_builddir)/pkgconfig"
+ if test "x$PKG_CONFIG_PATH" != "x"; then
+ GST_PKG_CONFIG_PATH="$GST_PKG_CONFIG_PATH:$PKG_CONFIG_PATH"
+ fi
+ AC_SUBST([GST_PKG_CONFIG_PATH])
+ AC_MSG_NOTICE([Using GST_PKG_CONFIG_PATH = $GST_PKG_CONFIG_PATH])
+])
--- /dev/null
+dnl -*- mode: autoconf -*-
+
+# serial 1
+
+dnl Usage:
+dnl GTK_DOC_CHECK([minimum-gtk-doc-version])
+AC_DEFUN([GTK_DOC_CHECK],
+[
+ AC_REQUIRE([PKG_PROG_PKG_CONFIG])
+ AC_BEFORE([AC_PROG_LIBTOOL],[$0])dnl setup libtool first
+ AC_BEFORE([AM_PROG_LIBTOOL],[$0])dnl setup libtool first
+
+ dnl check for tools we added during development
+ AC_PATH_PROG([GTKDOC_CHECK],[gtkdoc-check])
+ AC_PATH_PROGS([GTKDOC_REBASE],[gtkdoc-rebase],[true])
+ AC_PATH_PROG([GTKDOC_MKPDF],[gtkdoc-mkpdf])
+
+ dnl for overriding the documentation installation directory
+ AC_ARG_WITH([html-dir],
+ AS_HELP_STRING([--with-html-dir=PATH], [path to installed docs]),,
+ [with_html_dir='${datadir}/gtk-doc/html'])
+ HTML_DIR="$with_html_dir"
+ AC_SUBST([HTML_DIR])
+
+ dnl enable/disable documentation building
+ AC_ARG_ENABLE([gtk-doc],
+ AS_HELP_STRING([--enable-gtk-doc],
+ [use gtk-doc to build documentation [[default=no]]]),,
+ [enable_gtk_doc=no])
+
+ if test x$enable_gtk_doc = xyes; then
+ ifelse([$1],[],
+ [PKG_CHECK_EXISTS([gtk-doc],,
+ AC_MSG_ERROR([gtk-doc not installed and --enable-gtk-doc requested]))],
+ [PKG_CHECK_EXISTS([gtk-doc >= $1],,
+ AC_MSG_ERROR([You need to have gtk-doc >= $1 installed to build $PACKAGE_NAME]))])
+ dnl don't check for glib if we build glib
+ if test "x$PACKAGE_NAME" != "xglib"; then
+ dnl don't fail if someone does not have glib
+ PKG_CHECK_MODULES(GTKDOC_DEPS, glib-2.0 >= 2.10.0 gobject-2.0 >= 2.10.0,,)
+ fi
+ dnl don't rely on sed being pulled in implicitly. Fixes Solaris build.
+ if test -z "$SED"; then
+ AC_PROG_SED
+ fi
+ fi
+
+ AC_MSG_CHECKING([whether to build gtk-doc documentation])
+ AC_MSG_RESULT($enable_gtk_doc)
+
+ dnl enable/disable output formats
+ AC_ARG_ENABLE([gtk-doc-html],
+ AS_HELP_STRING([--enable-gtk-doc-html],
+ [build documentation in html format [[default=yes]]]),,
+ [enable_gtk_doc_html=yes])
+ AC_ARG_ENABLE([gtk-doc-pdf],
+ AS_HELP_STRING([--enable-gtk-doc-pdf],
+ [build documentation in pdf format [[default=no]]]),,
+ [enable_gtk_doc_pdf=no])
+
+ if test -z "$GTKDOC_MKPDF"; then
+ enable_gtk_doc_pdf=no
+ fi
+
+
+ AM_CONDITIONAL([ENABLE_GTK_DOC], [test x$enable_gtk_doc = xyes])
+ AM_CONDITIONAL([GTK_DOC_BUILD_HTML], [test x$enable_gtk_doc_html = xyes])
+ AM_CONDITIONAL([GTK_DOC_BUILD_PDF], [test x$enable_gtk_doc_pdf = xyes])
+ AM_CONDITIONAL([GTK_DOC_USE_LIBTOOL], [test -n "$LIBTOOL"])
+])
--- /dev/null
+dnl -*- mode: autoconf -*-
+dnl Copyright 2009 Johan Dahlin
+dnl
+dnl This file is free software; the author(s) gives unlimited
+dnl permission to copy and/or distribute it, with or without
+dnl modifications, as long as this notice is preserved.
+dnl
+
+# serial 1
+
+m4_define([_GOBJECT_INTROSPECTION_CHECK_INTERNAL],
+[
+ AC_BEFORE([AC_PROG_LIBTOOL],[$0])dnl setup libtool first
+ AC_BEFORE([AM_PROG_LIBTOOL],[$0])dnl setup libtool first
+ AC_BEFORE([LT_INIT],[$0])dnl setup libtool first
+
+ dnl enable/disable introspection
+ m4_if([$2], [require],
+ [dnl
+ enable_introspection=yes
+ ],[dnl
+ AC_ARG_ENABLE(introspection,
+ AS_HELP_STRING([--enable-introspection[=@<:@no/auto/yes@:>@]],
+ [Enable introspection for this build]),,
+ [enable_introspection=auto])
+ ])dnl
+
+ AC_MSG_CHECKING([for gobject-introspection])
+
+ dnl presence/version checking
+ AS_CASE([$enable_introspection],
+ [no], [dnl
+ found_introspection="no (disabled, use --enable-introspection to enable)"
+ ],dnl
+ [yes],[dnl
+ PKG_CHECK_EXISTS([gobject-introspection-1.0],,
+ AC_MSG_ERROR([gobject-introspection-1.0 is not installed]))
+ PKG_CHECK_EXISTS([gobject-introspection-1.0 >= $1],
+ found_introspection=yes,
+ AC_MSG_ERROR([You need to have gobject-introspection >= $1 installed to build AC_PACKAGE_NAME]))
+ ],dnl
+ [auto],[dnl
+ PKG_CHECK_EXISTS([gobject-introspection-1.0 >= $1], found_introspection=yes, found_introspection=no)
+ ],dnl
+ [dnl
+ AC_MSG_ERROR([invalid argument passed to --enable-introspection, should be one of @<:@no/auto/yes@:>@])
+ ])dnl
+
+ AC_MSG_RESULT([$found_introspection])
+
+ INTROSPECTION_SCANNER=
+ INTROSPECTION_COMPILER=
+ INTROSPECTION_GENERATE=
+ INTROSPECTION_GIRDIR=
+ INTROSPECTION_TYPELIBDIR=
+ if test "x$found_introspection" = "xyes"; then
+ INTROSPECTION_SCANNER=`$PKG_CONFIG --variable=g_ir_scanner gobject-introspection-1.0`
+ INTROSPECTION_COMPILER=`$PKG_CONFIG --variable=g_ir_compiler gobject-introspection-1.0`
+ INTROSPECTION_GENERATE=`$PKG_CONFIG --variable=g_ir_generate gobject-introspection-1.0`
+ INTROSPECTION_GIRDIR=`$PKG_CONFIG --variable=girdir gobject-introspection-1.0`
+ INTROSPECTION_TYPELIBDIR="$($PKG_CONFIG --variable=typelibdir gobject-introspection-1.0)"
+ INTROSPECTION_CFLAGS=`$PKG_CONFIG --cflags gobject-introspection-1.0`
+ INTROSPECTION_LIBS=`$PKG_CONFIG --libs gobject-introspection-1.0`
+ INTROSPECTION_MAKEFILE=`$PKG_CONFIG --variable=datadir gobject-introspection-1.0`/gobject-introspection-1.0/Makefile.introspection
+ fi
+ AC_SUBST(INTROSPECTION_SCANNER)
+ AC_SUBST(INTROSPECTION_COMPILER)
+ AC_SUBST(INTROSPECTION_GENERATE)
+ AC_SUBST(INTROSPECTION_GIRDIR)
+ AC_SUBST(INTROSPECTION_TYPELIBDIR)
+ AC_SUBST(INTROSPECTION_CFLAGS)
+ AC_SUBST(INTROSPECTION_LIBS)
+ AC_SUBST(INTROSPECTION_MAKEFILE)
+
+ AM_CONDITIONAL(HAVE_INTROSPECTION, test "x$found_introspection" = "xyes")
+])
+
+
+dnl Usage:
+dnl GOBJECT_INTROSPECTION_CHECK([minimum-g-i-version])
+
+AC_DEFUN([GOBJECT_INTROSPECTION_CHECK],
+[
+ _GOBJECT_INTROSPECTION_CHECK_INTERNAL([$1])
+])
+
+dnl Usage:
+dnl GOBJECT_INTROSPECTION_REQUIRE([minimum-g-i-version])
+
+
+AC_DEFUN([GOBJECT_INTROSPECTION_REQUIRE],
+[
+ _GOBJECT_INTROSPECTION_CHECK_INTERNAL([$1], [require])
+])
--- /dev/null
+dnl pkg-config-based checks for Orc
+
+dnl specific:
+dnl ORC_CHECK([REQUIRED_VERSION])
+
+AC_DEFUN([ORC_CHECK],
+[
+ ORC_REQ=ifelse([$1], , "0.4.6", [$1])
+
+ AC_ARG_ENABLE(orc,
+ AC_HELP_STRING([--enable-orc],[use Orc if installed]),
+ [case "${enableval}" in
+ auto) enable_orc=auto ;;
+ yes) enable_orc=yes ;;
+ no) enable_orc=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --enable-orc) ;;
+ esac
+ ],
+ [enable_orc=auto]) dnl Default value
+
+ if test "x$enable_orc" != "xno" ; then
+ PKG_CHECK_MODULES(ORC, orc-0.4 >= $ORC_REQ, [
+ AC_DEFINE(HAVE_ORC, 1, [Use Orc])
+ HAVE_ORC=yes
+ if test "x$ORCC" = "x" ; then
+ AC_MSG_CHECKING(for usable orcc)
+ ORCC=`$PKG_CONFIG --variable=orcc orc-0.4`
+ dnl check whether the orcc found by pkg-config can be run from the build environment
+ dnl if this is not the case (e.g. when cross-compiling) fall back to orcc from PATH
+ AS_IF([$ORCC --version 1> /dev/null 2> /dev/null], [], [ORCC=`which orcc`])
+ AC_MSG_RESULT($ORCC)
+ fi
+ AC_SUBST(ORCC)
+ ORCC_FLAGS="--compat $ORC_REQ"
+ AC_SUBST(ORCC_FLAGS)
+ AS_IF([test "x$ORCC" = "x"], [HAVE_ORCC=no], [HAVE_ORCC=yes])
+ ], [
+ if test "x$enable_orc" = "xyes" ; then
+ AC_MSG_ERROR([--enable-orc specified, but Orc >= $ORC_REQ not found])
+ fi
+ AC_DEFINE(DISABLE_ORC, 1, [Disable Orc])
+ HAVE_ORC=no
+ HAVE_ORCC=no
+ ])
+ else
+ AC_DEFINE(DISABLE_ORC, 1, [Disable Orc])
+ HAVE_ORC=no
+ HAVE_ORCC=no
+ fi
+ AM_CONDITIONAL(HAVE_ORC, [test "x$HAVE_ORC" = "xyes"])
+ AM_CONDITIONAL(HAVE_ORCC, [test "x$HAVE_ORCC" = "xyes"])
+
+]))
+
+AC_DEFUN([ORC_OUTPUT],
+[
+ if test "$HAVE_ORC" = yes ; then
+ printf "configure: *** Orc acceleration enabled.\n"
+ else
+ if test "x$enable_orc" = "xno" ; then
+ printf "configure: *** Orc acceleration disabled by --disable-orc. Slower code paths\n"
+ printf " will be used.\n"
+ else
+ printf "configure: *** Orc acceleration disabled. Requires Orc >= $ORC_REQ, which was\n"
+ printf " not found. Slower code paths will be used.\n"
+ fi
+ fi
+ printf "\n"
+])
+
--- /dev/null
+# pkg.m4 - Macros to locate and utilise pkg-config. -*- Autoconf -*-
+#
+# Copyright © 2004 Scott James Remnant <scott@netsplit.com>.
+#
+# This program is free software; you can redistribute it and/or modify
+# it under the terms of the GNU General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+# General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+#
+# As a special exception to the GNU General Public License, if you
+# distribute this file as part of a program that contains a
+# configuration script generated by Autoconf, you may include it under
+# the same distribution terms that you use for the rest of that program.
+
+# PKG_PROG_PKG_CONFIG([MIN-VERSION])
+# ----------------------------------
+AC_DEFUN([PKG_PROG_PKG_CONFIG],
+[m4_pattern_forbid([^_?PKG_[A-Z_]+$])
+m4_pattern_allow([^PKG_CONFIG(_PATH)?$])
+AC_ARG_VAR([PKG_CONFIG], [path to pkg-config utility])dnl
+if test "x$ac_cv_env_PKG_CONFIG_set" != "xset"; then
+ AC_PATH_TOOL([PKG_CONFIG], [pkg-config])
+fi
+if test -n "$PKG_CONFIG"; then
+ _pkg_min_version=m4_default([$1], [0.9.0])
+ AC_MSG_CHECKING([pkg-config is at least version $_pkg_min_version])
+ if $PKG_CONFIG --atleast-pkgconfig-version $_pkg_min_version; then
+ AC_MSG_RESULT([yes])
+ else
+ AC_MSG_RESULT([no])
+ PKG_CONFIG=""
+ fi
+
+fi[]dnl
+])# PKG_PROG_PKG_CONFIG
+
+# PKG_CHECK_EXISTS(MODULES, [ACTION-IF-FOUND], [ACTION-IF-NOT-FOUND])
+#
+# Check to see whether a particular set of modules exists. Similar
+# to PKG_CHECK_MODULES(), but does not set variables or print errors.
+#
+#
+# Similar to PKG_CHECK_MODULES, make sure that the first instance of
+# this or PKG_CHECK_MODULES is called, or make sure to call
+# PKG_CHECK_EXISTS manually
+# --------------------------------------------------------------
+AC_DEFUN([PKG_CHECK_EXISTS],
+[AC_REQUIRE([PKG_PROG_PKG_CONFIG])dnl
+if test -n "$PKG_CONFIG" && \
+ AC_RUN_LOG([$PKG_CONFIG --exists --print-errors "$1"]); then
+ m4_ifval([$2], [$2], [:])
+m4_ifvaln([$3], [else
+ $3])dnl
+fi])
+
+
+# _PKG_CONFIG([VARIABLE], [COMMAND], [MODULES])
+# ---------------------------------------------
+m4_define([_PKG_CONFIG],
+[if test -n "$PKG_CONFIG"; then
+ if test -n "$$1"; then
+ pkg_cv_[]$1="$$1"
+ else
+ PKG_CHECK_EXISTS([$3],
+ [pkg_cv_[]$1=`$PKG_CONFIG --[]$2 "$3" 2>/dev/null`],
+ [pkg_failed=yes])
+ fi
+else
+ pkg_failed=untried
+fi[]dnl
+])# _PKG_CONFIG
+
+# _PKG_SHORT_ERRORS_SUPPORTED
+# -----------------------------
+AC_DEFUN([_PKG_SHORT_ERRORS_SUPPORTED],
+[AC_REQUIRE([PKG_PROG_PKG_CONFIG])
+if $PKG_CONFIG --atleast-pkgconfig-version 0.20; then
+ _pkg_short_errors_supported=yes
+else
+ _pkg_short_errors_supported=no
+fi[]dnl
+])# _PKG_SHORT_ERRORS_SUPPORTED
+
+
+# PKG_CHECK_MODULES(VARIABLE-PREFIX, MODULES, [ACTION-IF-FOUND],
+# [ACTION-IF-NOT-FOUND])
+#
+#
+# Note that if there is a possibility the first call to
+# PKG_CHECK_MODULES might not happen, you should be sure to include an
+# explicit call to PKG_PROG_PKG_CONFIG in your configure.ac
+#
+#
+# --------------------------------------------------------------
+AC_DEFUN([PKG_CHECK_MODULES],
+[AC_REQUIRE([PKG_PROG_PKG_CONFIG])dnl
+AC_ARG_VAR([$1][_CFLAGS], [C compiler flags for $1, overriding pkg-config])dnl
+AC_ARG_VAR([$1][_LIBS], [linker flags for $1, overriding pkg-config])dnl
+
+pkg_failed=no
+AC_MSG_CHECKING([for $1])
+
+_PKG_CONFIG([$1][_CFLAGS], [cflags], [$2])
+_PKG_CONFIG([$1][_LIBS], [libs], [$2])
+
+m4_define([_PKG_TEXT], [Alternatively, you may set the environment variables $1[]_CFLAGS
+and $1[]_LIBS to avoid the need to call pkg-config.
+See the pkg-config man page for more details.])
+
+if test $pkg_failed = yes; then
+ _PKG_SHORT_ERRORS_SUPPORTED
+ if test $_pkg_short_errors_supported = yes; then
+ $1[]_PKG_ERRORS=`$PKG_CONFIG --short-errors --errors-to-stdout --print-errors "$2"`
+ else
+ $1[]_PKG_ERRORS=`$PKG_CONFIG --errors-to-stdout --print-errors "$2"`
+ fi
+ # Put the nasty error message in config.log where it belongs
+ echo "$$1[]_PKG_ERRORS" >&AS_MESSAGE_LOG_FD
+
+ ifelse([$4], , [AC_MSG_ERROR(dnl
+[Package requirements ($2) were not met:
+
+$$1_PKG_ERRORS
+
+Consider adjusting the PKG_CONFIG_PATH environment variable if you
+installed software in a non-standard prefix.
+
+_PKG_TEXT
+])],
+ [AC_MSG_RESULT([no])
+ $4])
+elif test $pkg_failed = untried; then
+ ifelse([$4], , [AC_MSG_FAILURE(dnl
+[The pkg-config script could not be found or is too old. Make sure it
+is in your PATH or set the PKG_CONFIG environment variable to the full
+path to pkg-config.
+
+_PKG_TEXT
+
+To get pkg-config, see <http://pkg-config.freedesktop.org/>.])],
+ [$4])
+else
+ $1[]_CFLAGS=$pkg_cv_[]$1[]_CFLAGS
+ $1[]_LIBS=$pkg_cv_[]$1[]_LIBS
+ AC_MSG_RESULT([yes])
+ ifelse([$3], , :, [$3])
+fi[]dnl
+])# PKG_CHECK_MODULES
--- /dev/null
+# -*- Mode: Python -*-
+# vi:si:et:sw=4:sts=4:ts=4
+
+"""
+use the output from gst-xmlinspect.py to mangle tmpl/*.sgml and
+insert/overwrite Short Description and Long Description
+"""
+
+# FIXME: right now it uses pygst and scans on its own;
+# we really should use inspect/*.xml instead since the result of
+# gst-xmlinspect.py is committed by the docs maintainer, who can be
+# expected to have pygst, but this step should be done for every docs build,
+# so no pygst allowed
+
+# read in inspect/*.xml
+# for every tmpl/element-(name).xml: mangle with details from element
+
+from __future__ import print_function, unicode_literals
+
+import glob
+import re
+import sys
+import os
+
+class Tmpl:
+ def __init__(self, filename):
+ self.filename = filename
+ self._sectionids = []
+ self._sections = {}
+
+ def read(self):
+ """
+ Read and parse the sections from the given file.
+ """
+ lines = open(self.filename).readlines()
+ matcher = re.compile("<!-- ##### SECTION (\S+) ##### -->\n")
+ id = None
+
+ for line in lines:
+ match = matcher.search(line)
+ if match:
+ id = match.expand("\\1")
+ self._sectionids.append(id)
+ self._sections[id] = []
+ else:
+ if not id:
+ sys.stderr.write(
+ "WARNING: line before a SECTION header: %s" % line)
+ else:
+ self._sections[id].append(line)
+
+ def get_section(self, id):
+ """
+ Get the content from the given section.
+ """
+ return self._sections[id]
+
+ def set_section(self, id, content):
+ """
+ Replace the given section id with the given content.
+ """
+ self._sections[id] = content
+
+ def output(self):
+ """
+ Return the output of the current template in the tmpl/*.sgml format.
+ """
+ lines = []
+ for id in self._sectionids:
+ lines.append("<!-- ##### SECTION %s ##### -->\n" % id)
+ for line in self._sections[id]:
+ lines.append(line)
+
+ return "".join(lines)
+
+ def write(self, backup=False):
+ """
+ Write out the template file again, backing up the previous one.
+ """
+ if backup:
+ target = self.filename + ".mangle.bak"
+ os.rename(self.filename, target)
+
+ handle = open(self.filename, "w")
+ handle.write(self.output())
+ handle.close()
+
+import xml.dom.minidom
+
+def get_elements(file):
+ elements = {}
+ doc = xml.dom.minidom.parse(file)
+
+ elem = None
+ for e in doc.childNodes:
+ if e.nodeType == e.ELEMENT_NODE and e.localName == 'plugin':
+ elem = e
+ break
+ if elem == None:
+ return None
+
+ elem2 = None
+ for e in elem.childNodes:
+ if e.nodeType == e.ELEMENT_NODE and e.localName == 'elements':
+ elem2 = e
+ break
+ if elem2 == None:
+ return None
+
+ elem = elem2
+
+ for e in elem.childNodes:
+ if e.nodeType == e.ELEMENT_NODE and e.localName == 'element':
+ name = None
+ description = None
+
+ for e2 in e.childNodes:
+ if e2.nodeType == e2.ELEMENT_NODE and e2.localName == 'name':
+ name = e2.childNodes[0].nodeValue.encode("UTF-8")
+ elif e2.nodeType == e2.ELEMENT_NODE and e2.localName == 'description':
+ if e2.childNodes:
+ description = e2.childNodes[0].nodeValue.encode("UTF-8")
+ else:
+ description = 'No description'
+
+ if name != None and description != None:
+ elements[name] = {'description': description}
+
+ return elements
+
+def main():
+ if not len(sys.argv) == 3:
+ sys.stderr.write('Please specify the inspect/ dir and the tmpl/ dir')
+ sys.exit(1)
+
+ inspectdir = sys.argv[1]
+ tmpldir = sys.argv[2]
+
+ # parse all .xml files; build map of element name -> short desc
+ #for file in glob.glob("inspect/plugin-*.xml"):
+ elements = {}
+ for file in glob.glob("%s/plugin-*.xml" % inspectdir):
+ elements.update(get_elements(file))
+
+ for file in glob.glob("%s/element-*.sgml" % tmpldir):
+ base = os.path.basename(file)
+ element = base[len("element-"):-len(".sgml")]
+ tmpl = Tmpl(file)
+ tmpl.read()
+ if element in elements.keys():
+ description = elements[element]['description']
+ tmpl.set_section("Short_Description", "%s\n\n" % description)
+
+ # put in an include if not yet there
+ line = '<include xmlns="http://www.w3.org/2003/XInclude" href="' + \
+ 'element-' + element + '-details.xml">' + \
+ '<fallback xmlns="http://www.w3.org/2003/XInclude" />' + \
+ '</include>\n'
+ section = tmpl.get_section("Long_Description")
+ if not section[0] == line:
+ section.insert(0, line)
+ tmpl.set_section("Long_Description", section)
+ tmpl.write()
+
+main()
--- /dev/null
+#
+# This is a makefile.am fragment to build Orc code.
+#
+# Define ORC_SOURCE and then include this file, such as:
+#
+# ORC_SOURCE=gstadderorc
+# include $(top_srcdir)/common/orc.mak
+#
+# This fragment will create tmp-orc.c and gstadderorc.h from
+# gstadderorc.orc.
+#
+# When 'make dist' is run at the top level, or 'make orc-update'
+# in a directory including this fragment, the generated source
+# files will be copied to $(ORC_SOURCE)-dist.[ch]. These files
+# should be checked in to git, since they are used if Orc is
+# disabled.
+#
+# Note that this file defines BUILT_SOURCES, so any later usage
+# of BUILT_SOURCES in the Makefile.am that includes this file
+# must use '+='.
+#
+
+
+EXTRA_DIST = $(ORC_SOURCE).orc
+
+ORC_NODIST_SOURCES = tmp-orc.c $(ORC_SOURCE).h
+BUILT_SOURCES = tmp-orc.c $(ORC_SOURCE).h
+
+
+orc-update: tmp-orc.c $(ORC_SOURCE).h
+ $(top_srcdir)/common/gst-indent tmp-orc.c
+ cp tmp-orc.c $(srcdir)/$(ORC_SOURCE)-dist.c
+ cp $(ORC_SOURCE).h $(srcdir)/$(ORC_SOURCE)-dist.h
+
+orcc_v_gen = $(orcc_v_gen_$(V))
+orcc_v_gen_ = $(orcc_v_gen_$(AM_DEFAULT_VERBOSITY))
+orcc_v_gen_0 = @echo " ORCC $@";
+
+cp_v_gen = $(cp_v_gen_$(V))
+cp_v_gen_ = $(cp_v_gen_$(AM_DEFAULT_VERBOSITY))
+cp_v_gen_0 = @echo " CP $@";
+
+if HAVE_ORCC
+tmp-orc.c: $(srcdir)/$(ORC_SOURCE).orc
+ $(orcc_v_gen)$(ORCC) $(ORCC_FLAGS) --implementation --include glib.h -o tmp-orc.c $(srcdir)/$(ORC_SOURCE).orc
+
+$(ORC_SOURCE).h: $(srcdir)/$(ORC_SOURCE).orc
+ $(orcc_v_gen)$(ORCC) $(ORCC_FLAGS) --header --include glib.h -o $(ORC_SOURCE).h $(srcdir)/$(ORC_SOURCE).orc
+else
+tmp-orc.c: $(srcdir)/$(ORC_SOURCE).orc $(srcdir)/$(ORC_SOURCE)-dist.c
+ $(cp_v_gen)cp $(srcdir)/$(ORC_SOURCE)-dist.c tmp-orc.c
+
+$(ORC_SOURCE).h: $(srcdir)/$(ORC_SOURCE).orc $(srcdir)/$(ORC_SOURCE)-dist.c
+ $(cp_v_gen)cp $(srcdir)/$(ORC_SOURCE)-dist.h $(ORC_SOURCE).h
+endif
+
+clean-local: clean-orc
+.PHONY: clean-orc
+clean-orc:
+ rm -f tmp-orc.c $(ORC_SOURCE).h
+
+dist-hook: dist-hook-orc
+.PHONY: dist-hook-orc
+
+# we try and copy updated orc -dist files below, but don't fail if it
+# doesn't work as the srcdir might not be writable
+dist-hook-orc: tmp-orc.c $(ORC_SOURCE).h
+ $(top_srcdir)/common/gst-indent tmp-orc.c
+ rm -f tmp-orc.c~
+ cmp -s tmp-orc.c $(srcdir)/$(ORC_SOURCE)-dist.c || \
+ cp tmp-orc.c $(srcdir)/$(ORC_SOURCE)-dist.c || true
+ cmp -s $(ORC_SOURCE).h $(srcdir)/$(ORC_SOURCE)-dist.h || \
+ cp $(ORC_SOURCE).h $(srcdir)/$(ORC_SOURCE)-dist.h || true
+ cp -p tmp-orc.c $(distdir)/$(ORC_SOURCE)-dist.c
+ cp -p $(ORC_SOURCE).h $(distdir)/$(ORC_SOURCE)-dist.h
+
--- /dev/null
+# include this at the end of $MODULE/ext/Makefile.am to force make to
+# build subdirectories in parallel when make -jN is used. We will end up
+# descending into all subdirectories a second time, but only after the first
+# (parallel) run has finished, so it should go right through the second time.
+
+.PHONY: independent-subdirs $(SUBDIRS)
+
+independent-subdirs: $(SUBDIRS)
+
+$(SUBDIRS):
+ $(MAKE) -C $@
+
+all-recursive: independent-subdirs
--- /dev/null
+<?xml version='1.0'?> <!--*- mode: xml -*-->
+
+<xsl:stylesheet
+ xmlns:xsl="http://www.w3.org/1999/XSL/Transform"
+ xmlns:exsl="http://exslt.org/common"
+ xmlns:str="http://exslt.org/strings"
+ extension-element-prefixes="exsl str"
+ version="1.0">
+<xsl:output method="xml" indent="yes"
+ doctype-public ="-//OASIS//DTD DocBook XML V4.1.2//EN"
+ doctype-system = "http://www.oasis-open.org/docbook/xml/4.1.2/docbookx.dtd"/>
+
+<xsl:param name="module" />
+
+ <xsl:template match="element">
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">
+ <xsl:element name="link">
+ <xsl:attribute name="linkend"><xsl:value-of select="$module" />-plugins-<xsl:value-of select="name"/></xsl:attribute>
+ <xsl:value-of select="name" />
+ </xsl:element>
+ </xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select="description" /></xsl:element>
+ </xsl:element>
+ </xsl:element>
+ <xsl:variable name="name"><xsl:copy-of select="name"/></xsl:variable>
+ <!-- here we write an element-(name)-details.xml file for the element -->
+ <exsl:document href="{concat ('xml/element-', $name, '-details.xml')}" method="xml" indent="yes">
+
+ <xsl:element name="refsynopsisdiv">
+ <xsl:element name="refsect2">
+ <xsl:element name="title">Element Information</xsl:element>
+ <xsl:element name="variablelist">
+
+ <!-- plugin name and link -->
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">plugin</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara">
+ <xsl:element name="link">
+ <xsl:attribute name="linkend">plugin-<xsl:value-of select="../../name"/></xsl:attribute>
+ <xsl:value-of select="../../name" />
+ </xsl:element>
+ </xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">author</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select="author" /></xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">class</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select="class" /></xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ </xsl:element> <!-- variablelist -->
+ </xsl:element> <!-- refsect2 -->
+
+ <xsl:element name="refsect2">
+ <xsl:element name="title">Element Pads</xsl:element>
+ <!-- process all caps -->
+ <xsl:for-each select="pads/caps">
+ <xsl:element name="variablelist">
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">name</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select="name" /></xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">direction</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select="direction" /></xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">presence</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select="presence" /></xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:for-each select='str:tokenize(details, ";")'>
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">
+ <xsl:if test="position()=1">details</xsl:if>
+ </xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select='.'/></xsl:element>
+ </xsl:element>
+ </xsl:element>
+ </xsl:for-each>
+
+ </xsl:element> <!-- variablelist -->
+
+ <!--xsl:element name="programlisting"><xsl:value-of select="details" /></xsl:element-->
+
+ </xsl:for-each>
+ </xsl:element> <!-- refsect2 -->
+ </xsl:element> <!-- refsynopsisdiv -->
+
+ </exsl:document>
+ </xsl:template>
+
+ <xsl:template match="plugin">
+ <xsl:element name="refentry">
+ <xsl:attribute name="id"><xsl:value-of select="$module" />-plugins-plugin-<xsl:value-of select="name"/></xsl:attribute>
+
+ <xsl:element name="refmeta">
+ <xsl:element name="refentrytitle">
+ <xsl:value-of select="name"/>
+ </xsl:element>
+ <xsl:element name="manvolnum">3</xsl:element>
+ <xsl:element name="refmiscinfo">FIXME Library</xsl:element>
+ </xsl:element> <!-- refmeta -->
+
+ <xsl:element name="refnamediv">
+ <xsl:element name="refname">
+ <xsl:value-of select="name"/>
+ </xsl:element>
+
+ <xsl:element name="refpurpose">
+ <xsl:element name="anchor">
+ <xsl:attribute name="id">plugin-<xsl:value-of select="name"/></xsl:attribute>
+ </xsl:element>
+ <xsl:value-of select="description"/>
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:element name="refsect1">
+ <xsl:element name="title">Plugin Information</xsl:element>
+ <xsl:element name="variablelist">
+
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">filename</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select="basename" /></xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">version</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select="version" /></xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">run-time license</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select="license" /></xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">package</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara"><xsl:value-of select="package" /></xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:element name="varlistentry">
+ <xsl:element name="term">origin</xsl:element>
+ <xsl:element name="listitem">
+ <xsl:element name="simpara">
+ <!-- only show origin as link if it starts with http -->
+ <xsl:choose>
+ <xsl:when test="substring(@href, 1, 4) = 'http'">
+ <xsl:element name="ulink">
+ <xsl:attribute name="url"><xsl:value-of select="origin" /></xsl:attribute>
+ <xsl:value-of select="origin" />
+ </xsl:element>
+ </xsl:when>
+ <xsl:otherwise>
+ <xsl:value-of select="origin" />
+ </xsl:otherwise>
+ </xsl:choose>
+ </xsl:element>
+ </xsl:element>
+ </xsl:element>
+
+ </xsl:element>
+ </xsl:element>
+
+ <xsl:element name="refsect1">
+ <xsl:element name="title">Elements</xsl:element>
+ <!-- process all elements -->
+ <xsl:element name="variablelist">
+ <xsl:apply-templates select="elements"/>
+ </xsl:element>
+ </xsl:element>
+
+ </xsl:element>
+
+ </xsl:template>
+
+ <!-- ignore -->
+ <xsl:template match="gst-plugin-paths" />
+
+</xsl:stylesheet>
--- /dev/null
+# rule to download the latest .po files
+download-po: $(top_srcdir)/common/download-translations
+ $(top_srcdir)/common/download-translations $(PACKAGE)
+
--- /dev/null
+# include this snippet to add a common release: target by using
+# include $(top_srcdir)/common/release.mak
+
+release: dist
+ @$(MAKE) $(PACKAGE)-$(VERSION).tar.xz.sha256sum
+ @echo
+ @echo "================================================================================================="
+ @echo "http://gstreamer.freedesktop.org/src/$(PACKAGE)/$(PACKAGE)-$(VERSION).tar.xz"
+ @cat $(PACKAGE)-$(VERSION).tar.xz.sha256sum
+ @echo "================================================================================================="
+ @if [ -d ~/releases/ ]; then \
+ cp -v $(PACKAGE)-$(VERSION).tar.xz ~/releases/; \
+ fi
+ @if [ -d ../www/data/src ]; then \
+ mv -v $(PACKAGE)-$(VERSION).tar.xz ../www/data/src/$(PACKAGE)/ ; \
+ mv -v $(PACKAGE)-$(VERSION).tar.xz.sha256sum ../www/data/src/$(PACKAGE)/ ; \
+ fi
+ @echo "================================================================================================="
+
+# generate sha256 sum files
+%.sha256sum: %
+ @sha256sum $< > $@
+
+# check that no marshal or enumtypes files are included
+# this in turn ensures that distcheck fails for missing .list files which is currently
+# shadowed when the corresponding .c and .h files are included.
+distcheck-hook:
+ @test "x" = "x`find $(distdir) -name \*-enumtypes.[ch] | grep -v win32`" && \
+ test "x" = "x`find $(distdir) -name \*-marshal.[ch]`" || \
+ ( echo "*** Leftover enumtypes or marshal files in the tarball." && \
+ echo "*** Make sure the following files are not disted:" && \
+ find $(distdir) -name \*-enumtypes.[ch] | grep -v win32 && \
+ find $(distdir) -name \*-marshal.[ch] && \
+ false )
--- /dev/null
+#!/usr/bin/python
+# -*- Mode: Python -*-
+# vi:si:et:sw=4:sts=4:ts=4
+
+"""
+parse, merge and write gstdoc-scanobj files
+"""
+
+from __future__ import print_function, unicode_literals
+
+import sys
+import os
+
+def debug(*args):
+ pass
+
+# OrderedDict class based on
+# http://aspn.activestate.com/ASPN/Cookbook/Python/Recipe/107747
+# Licensed under the Python License
+class OrderedDict(dict):
+ def __init__(self):
+ self._keys = []
+ dict.__init__(self)
+
+ def __delitem__(self, key):
+ dict.__delitem__(self, key)
+ self._keys.remove(key)
+
+ def __setitem__(self, key, item):
+ dict.__setitem__(self, key, item)
+ if key not in self._keys: self._keys.append(key)
+
+ def clear(self):
+ dict.clear(self)
+ self._keys = []
+
+ def copy(self):
+ dict = dict.copy(self)
+ dict._keys = self._keys[:]
+ return dict
+
+ def items(self):
+ return zip(self._keys, self.values())
+
+ def keys(self):
+ return self._keys
+
+ def popitem(self):
+ try:
+ key = self._keys[-1]
+ except IndexError:
+ raise KeyError('dictionary is empty')
+
+ val = self[key]
+ del self[key]
+
+ return (key, val)
+
+ def setdefault(self, key, failobj = None):
+ dict.setdefault(self, key, failobj)
+ if key not in self._keys: self._keys.append(key)
+
+ def update(self, dict):
+ dict.update(self, dict)
+ for key in dict.keys():
+ if key not in self._keys: self._keys.append(key)
+
+ def values(self):
+ return map(self.get, self._keys)
+
+class Object:
+ def __init__(self, name):
+ self._signals = OrderedDict()
+ self._args = OrderedDict()
+ self.name = name
+
+ def __repr__(self):
+ return "<Object %s>" % self.name
+
+ def add_signal(self, signal, overwrite=True):
+ if not overwrite and signal.name in self._signals:
+ raise IndexError("signal %s already in %r" % (signal.name, self))
+ self._signals[signal.name] = signal
+
+ def add_arg(self, arg, overwrite=True):
+ if not overwrite and arg.name in self._args:
+ raise IndexError("arg %s already in %r" % (arg.name, self))
+ self._args[arg.name] = arg
+
+class Docable:
+ def __init__(self, **kwargs):
+ for key in self.attrs:
+ setattr(self, key, kwargs[key])
+ self.dict = kwargs
+
+ def __repr__(self):
+ return "<%r %s>" % (str(self.__class__), self.name)
+
+class Signal(Docable):
+ attrs = ['name', 'returns', 'args']
+
+class Arg(Docable):
+ attrs = ['name', 'type', 'range', 'flags', 'nick', 'blurb', 'default']
+
+class GDoc:
+ def load_file(self, filename):
+ try:
+ lines = open(filename).readlines()
+ self.load_data("".join(lines))
+ except IOError:
+ print ("WARNING - could not read from %s" % filename)
+
+ def save_file(self, filename, backup=False):
+ """
+ Save the information to the given file if the file content changed.
+ """
+ olddata = None
+ try:
+ lines = open(filename).readlines()
+ olddata = "".join(lines)
+ except IOError:
+ print ("WARNING - could not read from %s" % filename)
+ newdata = self.get_data()
+ if olddata and olddata == newdata:
+ return
+
+ if olddata:
+ if backup:
+ os.rename(filename, filename + '.bak')
+
+ handle = open(filename, "w")
+ handle.write(newdata)
+ handle.close()
+
+class Signals(GDoc):
+ def __init__(self):
+ self._objects = OrderedDict()
+
+ def load_data(self, data):
+ """
+ Load the .signals lines, creating our list of objects and signals.
+ """
+ import re
+ smatcher = re.compile(
+ '(?s)' # make . match \n
+ '<SIGNAL>\n(.*?)</SIGNAL>\n'
+ )
+ nmatcher = re.compile(
+ '<NAME>'
+ '(?P<object>\S*)' # store object
+ '::'
+ '(?P<signal>\S*)' # store signal
+ '</NAME>'
+ )
+ rmatcher = re.compile(
+ '(?s)' # make . match \n
+ '<RETURNS>(?P<returns>\S*)</RETURNS>\n' # store returns
+ '(?P<args>.*)' # store args
+ )
+ for block in smatcher.findall(data):
+ nmatch = nmatcher.search(block)
+ if nmatch:
+ o = nmatch.group('object')
+ debug("Found object", o)
+ debug("Found signal", nmatch.group('signal'))
+ if o not in self._objects:
+ object = Object(o)
+ self._objects[o] = object
+
+ rmatch = rmatcher.search(block)
+ if rmatch:
+ dict = rmatch.groupdict().copy()
+ dict['name'] = nmatch.group('signal')
+ signal = Signal(**dict)
+ self._objects[o].add_signal(signal)
+
+ def get_data(self):
+ lines = []
+ for o in self._objects.values():
+ for s in o._signals.values():
+ block = """<SIGNAL>
+<NAME>%(object)s::%(name)s</NAME>
+<RETURNS>%(returns)s</RETURNS>
+%(args)s</SIGNAL>
+"""
+ d = s.dict.copy()
+ d['object'] = o.name
+ lines.append(block % d)
+
+ return "\n".join(lines) + '\n'
+
+class Args(GDoc):
+ def __init__(self):
+ self._objects = OrderedDict()
+
+ def load_data(self, data):
+ """
+ Load the .args lines, creating our list of objects and args.
+ """
+ import re
+ amatcher = re.compile(
+ '(?s)' # make . match \n
+ '<ARG>\n(.*?)</ARG>\n'
+ )
+ nmatcher = re.compile(
+ '<NAME>'
+ '(?P<object>\S*)' # store object
+ '::'
+ '(?P<arg>\S*)' # store arg
+ '</NAME>'
+ )
+ rmatcher = re.compile(
+ '(?s)' # make . match \n
+ '<TYPE>(?P<type>\S*)</TYPE>\n' # store type
+ '<RANGE>(?P<range>.*?)</RANGE>\n' # store range
+ '<FLAGS>(?P<flags>\S*)</FLAGS>\n' # store flags
+ '<NICK>(?P<nick>.*?)</NICK>\n' # store nick
+ '<BLURB>(?P<blurb>.*?)</BLURB>\n' # store blurb
+ '<DEFAULT>(?P<default>.*?)</DEFAULT>\n' # store default
+ )
+ for block in amatcher.findall(data):
+ nmatch = nmatcher.search(block)
+ if nmatch:
+ o = nmatch.group('object')
+ debug("Found object", o)
+ debug("Found arg", nmatch.group('arg'))
+ if o not in self._objects:
+ object = Object(o)
+ self._objects[o] = object
+
+ rmatch = rmatcher.search(block)
+ if rmatch:
+ dict = rmatch.groupdict().copy()
+ dict['name'] = nmatch.group('arg')
+ arg = Arg(**dict)
+ self._objects[o].add_arg(arg)
+ else:
+ print ("ERROR: could not match arg from block %s" % block)
+
+ def get_data(self):
+ lines = []
+ for o in self._objects.values():
+ for a in o._args.values():
+ block = """<ARG>
+<NAME>%(object)s::%(name)s</NAME>
+<TYPE>%(type)s</TYPE>
+<RANGE>%(range)s</RANGE>
+<FLAGS>%(flags)s</FLAGS>
+<NICK>%(nick)s</NICK>
+<BLURB>%(blurb)s</BLURB>
+<DEFAULT>%(default)s</DEFAULT>
+</ARG>
+"""
+ d = a.dict.copy()
+ d['object'] = o.name
+ lines.append(block % d)
+
+ return "\n".join(lines) + '\n'
+
+class SingleLine(GDoc):
+ def __init__(self):
+ self._objects = []
+
+ def load_data(self, data):
+ """
+ Load the .interfaces/.prerequisites lines, merge duplicates
+ """
+ # split data on '\n'
+ lines = data.splitlines();
+ # merge them into self._objects
+ for line in lines:
+ if line not in self._objects:
+ self._objects.append(line)
+
+ def get_data(self):
+ lines = sorted(self._objects)
+ return "\n".join(lines) + '\n'
+
+def main(argv):
+ modulename = None
+ try:
+ modulename = argv[1]
+ except IndexError:
+ sys.stderr.write('Please provide a documentation module name\n')
+ sys.exit(1)
+
+ signals = Signals()
+ signals.load_file(modulename + '.signals')
+ signals.load_file(modulename + '.signals.new')
+ signals.save_file(modulename + '.signals', backup=True)
+ os.unlink(modulename + '.signals.new')
+
+ args = Args()
+ args.load_file(modulename + '.args')
+ args.load_file(modulename + '.args.new')
+ args.save_file(modulename + '.args', backup=True)
+ os.unlink(modulename + '.args.new')
+
+ ifaces = SingleLine()
+ ifaces.load_file(modulename + '.interfaces')
+ ifaces.load_file(modulename + '.interfaces.new')
+ ifaces.save_file(modulename + '.interfaces', backup=True)
+ os.unlink(modulename + '.interfaces.new')
+
+ prereq = SingleLine()
+ prereq.load_file(modulename + '.prerequisites')
+ prereq.load_file(modulename + '.prerequisites.new')
+ prereq.save_file(modulename + '.prerequisites', backup=True)
+ os.unlink(modulename + '.prerequisites.new')
+
+main(sys.argv)
--- /dev/null
+# this snippet is to be included by both our docbook manuals
+# and gtk-doc API references
+
+# it adds an upload target to each of these dir's Makefiles
+
+# each Makefile.am should define the following variables:
+# - DOC: the base name of the documentation
+# (faq, manual, pwg, gstreamer, gstreamer-libs)
+# - FORMATS: the formats in which DOC is output
+# (html ps pdf)
+
+# if you want to use it, make sure your $HOME/.ssh/config file contains the
+# correct User entry for the Host entry for the DOC_SERVER
+
+# these variables define the location of the online docs
+DOC_SERVER = gstreamer.freedesktop.org
+DOC_BASE = /srv/gstreamer.freedesktop.org/www/data/doc
+DOC_URL = $(DOC_SERVER):$(DOC_BASE)
+
+upload: $(FORMATS)
+ @if echo $(FORMATS) | grep html > /dev/null; then \
+ echo "Preparing docs for upload (rebasing cross-references) ..." ; \
+ if test x$(builddir) != x$(srcdir); then \
+ echo "make upload can only be used if srcdir == builddir"; \
+ exit 1; \
+ fi; \
+ # gtkdoc-rebase sometimes gets confused, so reset everything to \
+ # local links before rebasing to online links \
+ gtkdoc-rebase --html-dir=$(builddir)/html 2>/dev/null 2>/dev/null ; \
+ rebase=`gtkdoc-rebase --verbose --online --html-dir=$(builddir)/html` ; \
+ echo "$$rebase" | grep -e "On-*line"; \
+ for req in glib gobject gstreamer gstreamer-libs gst-plugins-base-libs; do \
+ if ! ( echo "$$rebase" | grep -i -e "On-*line.*/$$req/" ); then \
+ echo "===============================================================================" ; \
+ echo " Could not determine online location for $$req docs. Cross-referencing will be " ; \
+ echo " broken, so not uploading. Make sure the library's gtk-doc documentation is " ; \
+ echo " installed somewhere in /usr/share/gtk-doc. " ; \
+ echo "===============================================================================" ; \
+ exit 1; \
+ fi; \
+ done; \
+ export SRC="$$SRC html"; \
+ fi; \
+ if echo $(FORMATS) | grep ps > /dev/null; then export SRC="$$SRC $(DOC).ps"; fi; \
+ if echo $(FORMATS) | grep pdf > /dev/null; then export SRC="$$SRC $(DOC).pdf"; fi; \
+ \
+ # upload releases to both X.Y/ and head/ subdirectories \
+ export DIR=$(DOC_BASE)/gstreamer/$(PACKAGE_VERSION_MAJOR).$(PACKAGE_VERSION_MINOR)/$(DOC); \
+ echo Uploading $$SRC to $(DOC_SERVER):$$DIR; \
+ ssh $(DOC_SERVER) mkdir -p $$DIR; \
+ rsync -rv -e ssh --delete $$SRC $(DOC_SERVER):$$DIR; \
+ ssh $(DOC_SERVER) chmod -R g+w $$DIR; \
+ \
+ export DIR=$(DOC_BASE)/gstreamer/head/$(DOC); \
+ echo Uploading $$SRC to $(DOC_SERVER):$$DIR; \
+ ssh $(DOC_SERVER) mkdir -p $$DIR; \
+ rsync -rv -e ssh --delete $$SRC $(DOC_SERVER):$$DIR; \
+ ssh $(DOC_SERVER) chmod -R g+w $$DIR; \
+ \
+ if echo $(FORMATS) | grep html > /dev/null; then \
+ echo "Un-preparing docs for upload (rebasing cross-references) ..." ; \
+ gtkdoc-rebase --html-dir=$(builddir)/html ; \
+ fi; \
+ echo Done
--- /dev/null
+# various tests to make sure we dist the win32 stuff (for MSVC builds) right
+
+# the MANIFEST contains all win32 related files that should be disted
+win32 = $(shell cat $(top_srcdir)/win32/MANIFEST)
+
+# wildcard is apparently not portable to other makes, hence the use of find
+# these are library .def files with the symbols to export
+win32defs = $(shell find $(top_srcdir)/win32/common -name '*.def')
+
+# wildcard is apparently not portable to other makes, hence the use of find
+# these are files that need to be disted with CRLF line endings:
+win32crlf = $(shell find $(top_srcdir)/win32 -name '*.dsw' -o -name '*.dsp')
+
+win32-debug:
+ @echo; \
+ echo win32 = $(win32); \
+ echo; \
+ echo win32defs = $(win32defs); \
+ echo; \
+ echo win32crlf = $(win32crlf); \
+ echo
+
+win32-check-crlf:
+ @echo Checking win32 files for CR LF line endings ...; \
+ fail=0 ; \
+ for each in $(win32crlf) ; do \
+ result=`perl -e 'print grep(/\r\n/,<>)' "$$each" | wc -l`; \
+ if test "$$result" = 0 ; then \
+ echo $$each must be fixed to have CRLF line endings ; \
+ fail=1; \
+ fi ; \
+ done ; \
+ exit $$fail
+
+# make sure all symbols we export on linux are defined in the win32 .def too
+# (don't care about other unixes for now, it's enough if it works on one of
+# the linux build bots; we assume .so )
+check-exports:
+ @fail=0 ; \
+ for l in $(win32defs); do \
+ libbase=`basename "$$l" ".def"`; \
+ libso=`find "$(top_builddir)" -name "$$libbase-@GST_API_VERSION@.so" | grep -v /_build/ | head -n1`; \
+ libdef="$(top_srcdir)/win32/common/$$libbase.def"; \
+ if test "x$$libso" != "x"; then \
+ echo Checking symbols in $$libso; \
+ if ! ($(top_srcdir)/common/check-exports $$libdef $$libso) ; then \
+ fail=1; \
+ fi; \
+ fi; \
+ done ; \
+ if test $$fail != 0; then \
+ echo '-----------------------------------------------------------'; \
+ echo 'Run this to update the .def files:'; \
+ echo 'make update-exports'; \
+ echo '-----------------------------------------------------------'; \
+ fi; \
+ exit $$fail
+
+update-exports:
+ make check-exports 2>&1 | patch -p1
+ git add win32/common/libgst*.def
+ git diff --cached -- win32/common/
+ echo '^^^--- updated and staged changes above'
+
+# complain about nonportable printf format strings (%lld, %llu, %zu etc.)
+check-nonportable-print-format:
+ @fail=0 ; \
+ loc=`find "$(top_srcdir)" -name '*.c' | xargs grep -n -e '%[0-9]*ll[udx]' -e '%[0-9]*z[udx]'`; \
+ if test "x$$loc" != "x"; then \
+ echo "Please fix the following print format strings:" ; \
+ find "$(top_srcdir)" -name '*.c' | xargs grep -n -e '%[0-9]*ll[udx]' -e '%[0-9]*z[udx]'; \
+ fail=1; \
+ fi; \
+ exit $$fail
+
+dist-hook: check-exports win32-check-crlf
+
+
--- /dev/null
+AC_PREREQ(2.62)
+dnl initialize autoconf
+dnl when going to/from release please set the nano (fourth number) right !
+dnl releases only do Wall, cvs and prerelease does Werror too
+AC_INIT([GStreamer RTSP Server Library], [1.4.5],
+ [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
+ [gst-rtsp-server])
+AG_GST_INIT
+
+dnl initialize automake
+AM_INIT_AUTOMAKE([-Wno-portability 1.11 no-dist-gzip dist-xz tar-ustar subdir-objects])
+
+dnl define PACKAGE_VERSION_* variables
+AS_VERSION
+
+dnl check if this is a release version
+AS_NANO(GST_GIT="no", GST_GIT="yes")
+
+dnl can autoconf find the source ?
+AC_CONFIG_SRCDIR([gst/rtsp-server/rtsp-server.c])
+
+dnl define the output header for config
+AC_CONFIG_HEADERS([config.h])
+
+dnl AM_MAINTAINER_MODE only provides the option to configure to enable it
+AM_MAINTAINER_MODE([enable])
+
+dnl sets host_* variables
+AC_CANONICAL_HOST
+
+dnl use pretty build output with automake >= 1.11
+m4_ifdef([AM_SILENT_RULES],[AM_SILENT_RULES([yes])],
+ [AM_DEFAULT_VERBOSITY=1
+ AC_SUBST(AM_DEFAULT_VERBOSITY)])
+
+dnl our libraries and install dirs use major.minor as a version
+dnl GST_API_VERSION=$PACKAGE_VERSION_MAJOR.$PACKAGE_VERSION_MINOR
+dnl we override it here if we need to for the release candidate of new series
+GST_API_VERSION=1.0
+AC_SUBST(GST_API_VERSION)
+
+dnl CURRENT, REVISION, AGE
+dnl - library source changed -> increment REVISION
+dnl - interfaces added/removed/changed -> increment CURRENT, REVISION = 0
+dnl - interfaces added -> increment AGE
+dnl - interfaces removed -> AGE = 0
+dnl
+dnl Keep CURRENT as MINOR * 100 + MICRO
+dnl Ex : 1.0.0 => 0
+dnl 1.0.3 => 3
+dnl 1.1.0 => 100
+dnl 1.2.5 => 205
+dnl 1.10.9 (who knows) => 1009
+dnl
+dnl sets GST_LT_LDFLAGS
+AS_LIBTOOL(GST, 405, 0, 405)
+
+dnl *** required versions of GStreamer stuff ***
+GST_REQ=1.4.0
+GSTPB_REQ=1.4.0
+GSTPG_REQ=1.4.0
+GSTPD_REQ=1.4.0
+
+dnl *** autotools stuff ****
+
+dnl allow for different autotools
+AS_AUTOTOOLS_ALTERNATE
+
+dnl Add parameters for aclocal
+AC_SUBST(ACLOCAL_AMFLAGS, "-I m4 -I common/m4")
+AC_CONFIG_MACRO_DIR([m4])
+
+dnl set up gettext
+dnl the version check needs to stay here because autopoint greps for it
+#AM_GNU_GETTEXT_VERSION([0.17])
+#AM_GNU_GETTEXT([external])
+#AG_GST_GETTEXT([gst-rtsp-server-$GST_API_VERSION])
+
+dnl *** check for arguments to configure ***
+
+AG_GST_ARG_DISABLE_FATAL_WARNINGS
+
+AG_GST_ARG_DEBUG
+AG_GST_ARG_VALGRIND
+AG_GST_ARG_GCOV
+AG_GST_ARG_WITH_PKG_CONFIG_PATH
+AG_GST_ARG_WITH_PACKAGE_NAME
+AG_GST_ARG_WITH_PACKAGE_ORIGIN
+
+AG_GST_PKG_CONFIG_PATH
+
+AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO([$PACKAGE_VERSION_NANO],
+ ["${srcdir}/gst-rtsp-server.doap"],
+ [$PACKAGE_VERSION_MAJOR.$PACKAGE_VERSION_MINOR.$PACKAGE_VERSION_MICRO])
+
+dnl building of tests
+AC_ARG_ENABLE(tests,
+ AS_HELP_STRING([--disable-tests],[disable building test apps]),
+ [
+ case "${enableval}" in
+ yes) BUILD_TESTS=yes ;;
+ no) BUILD_TESTS=no ;;
+ *) AC_MSG_ERROR(bad value ${enableval} for --disable-tests) ;;
+ esac
+ ],
+[BUILD_TESTS=yes]) dnl Default value
+AM_CONDITIONAL(BUILD_TESTS, test "x$BUILD_TESTS" = "xyes")
+
+dnl *** checks for platform ***
+
+dnl * hardware/architecture *
+
+dnl *** checks for programs ***
+
+dnl find a compiler
+AC_PROG_CC
+AC_PROG_CC_STDC
+
+dnl check if the compiler supports '-c' and '-o' options
+AM_PROG_CC_C_O
+
+dnl find an assembler
+AM_PROG_AS
+
+AC_PATH_PROG(VALGRIND_PATH, valgrind, no)
+AM_CONDITIONAL(HAVE_VALGRIND, test ! "x$VALGRIND_PATH" = "xno")
+
+dnl check for gobject-introspection
+GOBJECT_INTROSPECTION_CHECK([1.31.1])
+
+dnl check for documentation tools
+AG_GST_DOCBOOK_CHECK
+GTK_DOC_CHECK([1.12])
+
+dnl *** checks for libraries ***
+
+dnl *** checks for header files ***
+
+dnl *** checks for types/defines ***
+
+dnl *** checks for structures ***
+
+dnl *** checks for compiler characteristics ***
+
+dnl *** checks for library functions ***
+
+dnl *** checks for dependancy libraries ***
+
+dnl GLib is required
+GLIB_REQ=2.32.0
+AC_SUBST([GLIB_REQ])
+AG_GST_GLIB_CHECK([$GLIB_REQ])
+
+dnl checks for gstreamer
+dnl uninstalled is selected preferentially -- see pkg-config(1)
+AG_GST_CHECK_GST($GST_API_VERSION, [$GST_REQ], [yes])
+
+GST_TOOLS_DIR=`$PKG_CONFIG --variable=toolsdir gstreamer-$GST_API_VERSION`
+if test -z $GST_TOOLS_DIR; then
+ AC_MSG_ERROR([no tools dir defined in GStreamer pkg-config file; core upgrade needed.])
+fi
+AC_SUBST(GST_TOOLS_DIR)
+
+GST_PLUGINS_DIR=`$PKG_CONFIG gstreamer-$GST_API_VERSION --variable pluginsdir`
+AC_SUBST(GST_PLUGINS_DIR)
+AC_MSG_NOTICE(Using GStreamer Core Plugins in $GST_PLUGINS_DIR)
+
+AG_GST_CHECK_GST_BASE($GST_API_VERSION, [$GST_REQ], [yes])
+
+AG_GST_CHECK_GST_PLUGINS_BASE($GST_API_VERSION, [$GSTPB_REQ], [yes])
+GSTPB_PLUGINS_DIR=`$PKG_CONFIG gstreamer-plugins-base-$GST_API_VERSION --variable pluginsdir`
+AC_SUBST(GSTPB_PLUGINS_DIR)
+AC_MSG_NOTICE(Using GStreamer Base Plugins in $GSTPB_PLUGINS_DIR)
+
+AG_GST_CHECK_GST_PLUGINS_GOOD($GST_API_VERSION, [$GSTPG_REQ], [yes])
+GSTPG_PLUGINS_DIR=`$PKG_CONFIG gstreamer-plugins-good-$GST_API_VERSION --variable pluginsdir`
+AC_SUBST(GSTPG_PLUGINS_DIR)
+AC_MSG_NOTICE(Using GStreamer Good Plugins in $GSTPG_PLUGINS_DIR)
+
+AG_GST_CHECK_GST_PLUGINS_BAD($GST_API_VERSION, [$GSTPD_REQ], [yes])
+GSTPD_PLUGINS_DIR=`$PKG_CONFIG gstreamer-plugins-bad-$GST_API_VERSION --variable pluginsdir`
+AC_SUBST(GSTPD_PLUGINS_DIR)
+AC_MSG_NOTICE(Using GStreamer Bad Plugins in $GSTPD_PLUGINS_DIR)
+
+AG_GST_CHECK_GST_CHECK($GST_API_VERSION, [$GST_REQ], no)
+AM_CONDITIONAL(HAVE_CHECK, test "x$HAVE_GST_CHECK" = "xyes")
+
+dnl Check for -Bsymbolic-functions linker flag used to avoid
+dnl intra-library PLT jumps, if available.
+AC_ARG_ENABLE(Bsymbolic,
+ [AS_HELP_STRING([--disable-Bsymbolic],[avoid linking with -Bsymbolic])],,
+ [SAVED_LDFLAGS="${LDFLAGS}"
+ AC_MSG_CHECKING([for -Bsymbolic-functions linker flag])
+ LDFLAGS=-Wl,-Bsymbolic-functions
+ AC_LINK_IFELSE([AC_LANG_PROGRAM([[]], [[int main (void) { return 0; }]])],[
+ AC_MSG_RESULT(yes)
+ enable_Bsymbolic=yes],[
+ AC_MSG_RESULT(no)
+ enable_Bsymbolic=no])
+ LDFLAGS="${SAVED_LDFLAGS}"])
+
+dnl *** set variables based on configure arguments ***
+
+dnl set license and copyright notice
+GST_LICENSE="LGPL"
+AC_DEFINE_UNQUOTED(GST_LICENSE, "$GST_LICENSE", [GStreamer license])
+AC_SUBST(GST_LICENSE)
+
+dnl set location of plugin directory
+AG_GST_SET_PLUGINDIR
+
+dnl set release date/time (and check that release version is in doap file)
+AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO([$PACKAGE_VERSION_NANO],
+ ["${srcdir}/gst-rtsp-server.doap"],
+ [$PACKAGE_VERSION_MAJOR.$PACKAGE_VERSION_MINOR.$PACKAGE_VERSION_MICRO])
+
+# set by AG_GST_PARSE_SUBSYSTEM_DISABLES above
+dnl make sure it doesn't complain about unused variables if debugging is disabled
+NO_WARNINGS=""
+AG_GST_CHECK_GST_DEBUG_DISABLED([NO_WARNINGS="-Wno-unused"], [NO_WARNINGS=""])
+
+dnl define an ERROR_CFLAGS Makefile variable
+AG_GST_SET_ERROR_CFLAGS($FATAL_WARNINGS, [-Wmissing-declarations -Wmissing-prototypes -Wredundant-decls -Wundef -Wwrite-strings -Wformat-nonliteral -Wformat-security -Wold-style-definition -Winit-self -Wmissing-include-dirs -Waddress -Waggregate-return -Wno-multichar -Wnested-externs $NO_WARNINGS])
+
+dnl define correct level for debugging messages
+AG_GST_SET_LEVEL_DEFAULT($GST_GIT)
+
+dnl used in examples
+AG_GST_DEFAULT_ELEMENTS
+
+dnl *** finalize CFLAGS, LDFLAGS, LIBS
+
+dnl Overview:
+dnl GST_OPTION_CFLAGS: common flags for profiling, debugging, errors, ...
+dnl GST_*: flags shared by built objects to link against GStreamer
+dnl GST_ALL_LDFLAGS: linker flags shared by all
+dnl GST_LIB_LDFLAGS: additional linker flags for all libaries
+dnl GST_LT_LDFLAGS: library versioning of our libraries
+dnl GST_PLUGIN_LDFLAGS: flags to be used for all plugins
+
+dnl GST_OPTION_CFLAGS
+if test "x$USE_DEBUG" = xyes; then
+ PROFILE_CFLAGS="-g"
+fi
+AC_SUBST(PROFILE_CFLAGS)
+
+# GST_DISABLE_DEPRECATED: hide the visibility of deprecated
+# functionality from the API that gstreamer uses
+# GST_REMOVE_DEPRECATED: don't compile deprecated functionality (breaks ABI)
+if test "x$PACKAGE_VERSION_NANO" = "x1"; then
+ dnl Define _only_ when compiling from git (not for pre-releases or releases)
+ DEPRECATED_CFLAGS="-DGST_DISABLE_DEPRECATED"
+else
+ DEPRECATED_CFLAGS=""
+fi
+AC_SUBST(DEPRECATED_CFLAGS)
+
+dnl every flag in GST_OPTION_CFLAGS can be overridden at make time
+GST_OPTION_CFLAGS="\$(WARNING_CFLAGS) \$(ERROR_CFLAGS) \$(DEBUG_CFLAGS) \$(PROFILE_CFLAGS) \$(GCOV_CFLAGS) \$(OPT_CFLAGS) \$(DEPRECATED_CFLAGS)"
+AC_SUBST(GST_OPTION_CFLAGS)
+
+dnl FIXME: do we want to rename to GST_ALL_* ?
+dnl prefer internal headers to already installed ones
+dnl add GST_OPTION_CFLAGS, but overridable
+GST_CFLAGS="$GST_CFLAGS \$(GST_OPTION_CFLAGS)"
+AC_SUBST(GST_CFLAGS)
+AC_SUBST(GST_LIBS)
+
+dnl GST_ALL_*
+dnl vars common to for all internal objects (core libs, elements, applications)
+dnl CFLAGS:
+dnl - src and build dirs need to be added because every piece that gets built
+dnl will need the GStreamer source and generated headers
+GST_ALL_CFLAGS="-I\$(top_srcdir) -I\$(top_builddir) $GST_PLUGINS_BASE_CFLAGS $GST_CFLAGS \$(GST_OPTION_CFLAGS)"
+AC_SUBST([GST_ALL_CFLAGS])
+
+dnl FIXME: check if LTLIBINTL is needed everywhere
+dnl I presume it is given that it contains the symbols that _() stuff maps to
+GST_ALL_LIBS="$GST_LIBS $LTLIBINTL \$(GCOV_LIBS)"
+AC_SUBST([GST_ALL_LIBS])
+
+dnl LDFLAGS really should only contain flags, not libs - they get added before
+dnl whatevertarget_LIBS and -L flags here affect the rest of the linking
+GST_ALL_LDFLAGS="-no-undefined"
+if test "x${enable_Bsymbolic}" = "xyes"; then
+ GST_ALL_LDFLAGS="$GST_ALL_LDFLAGS -Wl,-Bsymbolic-functions"
+fi
+
+AC_SUBST(GST_ALL_LDFLAGS)
+
+dnl GST_LIB_LDFLAGS
+dnl linker flags shared by all libraries
+dnl LDFLAGS modifier defining exported symbols from built libraries
+GST_LIB_LDFLAGS="-export-symbols-regex \^[_]?\(gst_\|Gst\|GST_\).*"
+AC_SUBST(GST_LIB_LDFLAGS)
+
+dnl GST_OBJ_*
+dnl default vars for all internal objects built on libgstrtspserver
+dnl includes GST_ALL_*
+GST_OBJ_CFLAGS="\$(GST_ALL_CFLAGS)"
+AC_SUBST([GST_OBJ_CFLAGS])
+GST_OBJ_LIBS="\$(top_builddir)/gst/rtsp-server/libgstrtspserver-$GST_API_VERSION.la \$(GST_ALL_LIBS)"
+AC_SUBST([GST_OBJ_LIBS])
+
+PKG_CHECK_MODULES(LIBCGROUP, libcgroup >= 0.26, HAVE_LIBCGROUP="yes", HAVE_LIBCGROUP="no")
+AC_SUBST(LIBCGROUP_CFLAGS)
+AC_SUBST(LIBCGROUP_LIBS)
+AM_CONDITIONAL(HAVE_LIBCGROUP, test "x$HAVE_LIBCGROUP" = "xyes")
+
+dnl this really should only contain flags, not libs - they get added before
+dnl whatevertarget_LIBS and -L flags here affect the rest of the linking
+
+dnl *** output files ***
+
+dnl keep this alphabetic per directory, please
+AC_CONFIG_FILES([
+Makefile
+common/Makefile
+common/m4/Makefile
+gst/Makefile
+gst/rtsp-server/Makefile
+examples/Makefile
+tests/Makefile
+tests/check/Makefile
+pkgconfig/Makefile
+pkgconfig/gstreamer-rtsp-server.pc
+pkgconfig/gstreamer-rtsp-server-uninstalled.pc
+docs/Makefile
+docs/version.entities
+docs/libs/Makefile
+])
+AC_OUTPUT
+
+echo "
+
+Configuration
+ Version : ${VERSION}
+ Source code location : ${srcdir}
+ Prefix : ${prefix}
+ Compiler : ${CC}
+ CGroups example : ${HAVE_LIBCGROUP}
+
+gst-rtsp-server configured. Type 'make' to build.
+"
--- /dev/null
+if ENABLE_GTK_DOC
+DOCS_SUBDIRS = libs
+else
+DOCS_SUBDIRS =
+endif
+
+SUBDIRS = $(DOCS_SUBDIRS)
+DIST_SUBDIRS = libs
+
+EXTRA_DIST = \
+ version.entities.in
+
+upload:
+ @if test "x$(SUBDIRS)" != x; then \
+ for a in $(SUBDIRS); do \
+ cd $$a; \
+ make upload; \
+ cd ..; \
+ done; \
+ fi
--- /dev/null
+# Makefile.in generated by automake 1.14.1 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994-2013 Free Software Foundation, Inc.
+
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+VPATH = @srcdir@
+am__is_gnu_make = test -n '$(MAKEFILE_LIST)' && test -n '$(MAKELEVEL)'
+am__make_running_with_option = \
+ case $${target_option-} in \
+ ?) ;; \
+ *) echo "am__make_running_with_option: internal error: invalid" \
+ "target option '$${target_option-}' specified" >&2; \
+ exit 1;; \
+ esac; \
+ has_opt=no; \
+ sane_makeflags=$$MAKEFLAGS; \
+ if $(am__is_gnu_make); then \
+ sane_makeflags=$$MFLAGS; \
+ else \
+ case $$MAKEFLAGS in \
+ *\\[\ \ ]*) \
+ bs=\\; \
+ sane_makeflags=`printf '%s\n' "$$MAKEFLAGS" \
+ | sed "s/$$bs$$bs[$$bs $$bs ]*//g"`;; \
+ esac; \
+ fi; \
+ skip_next=no; \
+ strip_trailopt () \
+ { \
+ flg=`printf '%s\n' "$$flg" | sed "s/$$1.*$$//"`; \
+ }; \
+ for flg in $$sane_makeflags; do \
+ test $$skip_next = yes && { skip_next=no; continue; }; \
+ case $$flg in \
+ *=*|--*) continue;; \
+ -*I) strip_trailopt 'I'; skip_next=yes;; \
+ -*I?*) strip_trailopt 'I';; \
+ -*O) strip_trailopt 'O'; skip_next=yes;; \
+ -*O?*) strip_trailopt 'O';; \
+ -*l) strip_trailopt 'l'; skip_next=yes;; \
+ -*l?*) strip_trailopt 'l';; \
+ -[dEDm]) skip_next=yes;; \
+ -[JT]) skip_next=yes;; \
+ esac; \
+ case $$flg in \
+ *$$target_option*) has_opt=yes; break;; \
+ esac; \
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+
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+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
--- /dev/null
+README
+------
+
+(Last updated on Mon 15 jul 2013, version 0.11.90.1)
+
+This HOWTO describes the basic usage of the GStreamer RTSP libraries and how you
+can build simple server applications with it.
+
+* General
+
+ The server relies heavily on the RTSP infrastructure of GStreamer. This includes
+ all of the media acquisition, decoding, encoding, payloading and UDP/TCP
+ streaming. We use the rtpbin element for all the session management. Most of
+ the RTSP message parsing and construction in the server is done using the RTSP
+ library that comes with gst-plugins-base.
+
+ The result is that the server is rather small (a few 11000 lines of code) and easy
+ to understand and extend. In its current state of development, things change
+ fast, API and ABI are unstable. We encourage people to use it for their various
+ use cases and participate by suggesting changes/features.
+
+ Most of the server is built as a library containing a bunch of GObject objects
+ that provide reasonable default functionality but has a fair amount of hooks
+ to override the default behaviour.
+
+ The server currently integrates with the glib mainloop nicely. It's currently
+ not meant to be used in high-load scenarios and because no security audit has
+ been done, you should probably not put it on a public IP address.
+
+* Initialisation
+
+ You need to initialize GStreamer before using any of the RTSP server functions.
+
+ #include <gst/gst.h>
+
+ int
+ main (int argc, char *argv[])
+ {
+ gst_init (&argc, &argv);
+
+ ...
+ }
+
+ The server itself currently does not have any specific initialisation function
+ but that might change in the future.
+
+
+* Creating the server
+
+ The first thing you want to do is create a new GstRTSPServer object. This object
+ will handle all the new client connections to your server once it is added to a
+ GMainLoop. You can create a new server object like this:
+
+ #include <gst/rtsp-server/rtsp-server.h>
+
+ GstRTSPServer *server;
+
+ server = gst_rtsp_server_new ();
+
+ The server will by default listen on port 8554 for new connections. This can be
+ changed by calling gst_rtsp_server_set_service() or with the 'service' GObject
+ property. This makes it possible to run multiple server instances listening on
+ multiple ports on one machine.
+
+ We can make the server start listening on its default port by attaching it to a
+ mainloop. The following example shows how this is done and will start a server
+ on the default 8554 port. For any request we make, we will get a NOT_FOUND
+ error code because we need to configure more things before the server becomes
+ useful.
+
+ #include <gst/gst.h>
+ #include <gst/rtsp-server/rtsp-server.h>
+
+ int
+ main (int argc, char *argv[])
+ {
+ GstRTSPServer *server;
+ GMainLoop *loop;
+
+ gst_init (&argc, &argv);
+
+ server = gst_rtsp_server_new ();
+
+ /* make a mainloop for the default context */
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_main_loop_run (loop);
+ }
+
+ The server manages four other objects: GstRTSPSessionPool,
+ GstRTSPMountPoints, GstRTSPAuth and GstRTSPThreadPool.
+
+ The GstRTSPSessionPool is an object that keeps track of all the active sessions
+ in the server. A session will usually be kept for each client that performed a
+ SETUP request for a certain media stream. It contains the configuration that
+ the client negotiated with the server to receive the particular stream, ie. the
+ transport used and port pairs for UDP along with the state of the streaming.
+ The default implementation of the session pool is usually sufficient but
+ alternative implementation can be used by the server.
+
+ The GstRTSPMountPoints object is more interesting and needs more configuration
+ before the server object is useful. This object manages the mapping from a
+ request URL to a specific stream and its configuration. We explain in the next
+ topic how to configure this object.
+
+ GstRTSPAuth is an object that authenticates users and authorizes actions
+ performed by users. By default, a server does not have a GstRTSPAuth object and
+ thus does not try to perform any authentication or authorization.
+
+ GstRTSPThreadPool manages the threads used for client connections and media
+ pipelines. The server has a default implementation of a threadpool that should
+ be sufficient in most cases.
+
+
+* Making url mount points
+
+ Next we need to define what media is attached to a particular URL. What we want
+ to achieve is that when the user asks our server for a specific URL, say /test,
+ that we create (or reuse) a GStreamer pipeline that produces one or more RTP
+ streams.
+
+ The object that can create such pipeline is called a GstRTSPMediaFactory object.
+ The default implementation of GstRTSPMediaFactory allows you to easily create
+ GStreamer pipelines using the gst-launch syntax. It is possible to create a
+ GstRTSPMediaFactory subclass that uses different methods for constructing
+ pipelines.
+
+ The default GstRTSPMediaFactory can be configured with a gst-launch line that
+ produces a toplevel bin (use '(' and ')' around the pipeline description to
+ force a toplevel GstBin instead of the default GstPipeline toplevel element).
+ The pipeline description should contain elements named payN, one for each
+ stream (ex. pay0, pay1, ...). Also, for increased compatibility each stream
+ should have a different payload type which can be configured on the payloader.
+
+ The following code snippet illustrates how to create a media factory that
+ creates an RTP feed of an H264 encoded test video signal.
+
+ GstRTSPMediaFactory *factory;
+
+ factory = gst_rtsp_media_factory_new ();
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! x264enc ! rtph264pay pt=96 name=pay0 )");
+
+ Now that we have the media factory, we can attach it to a specific url. To do
+ this we get the default GstRTSPMountPoints from our server and add the url to
+ factory mount points to it like this:
+
+ GstRTSPMountPoints *mounts;
+
+ ...create server..create factory..
+
+ /* get the default mount points from the server */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* attach the video test signal to the "/test" URL */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+ g_object_unref (mounts);
+
+ When starting the server now and directing an RTP client to the URL (like with
+ vlc, mplayer or gstreamer):
+
+ rtsp://localhost:8554/test
+
+ a test signal will be streamed to the client. The full example code can be
+ found in the examples/test-readme.c file.
+
+ Note that by default the factory will create a new pipeline for each client. If
+ you want to share a pipeline between clients, use
+ gst_rtsp_media_factory_set_shared().
+
+
+* more on GstRTSPMediaFactory
+
+ The GstRTSPMediaFactory is responsible for creating and caching GstRTSPMedia
+ objects.
+
+ A freshly created GstRTSPMedia object from the factory initially only contains a
+ GstElement containing the elements to produce the RTP streams for the media and
+ a GPtrArray of GstRTSPStream objects describing the payloader and its source
+ pad. The media is unprepared in this state.
+
+ Usually the url will determine what kind of pipeline should be created. You can
+ for example use query parameters to configure certain parts of the pipeline or
+ select encoders and payloaders based on some url pattern.
+
+ When dealing with a live stream from, for example, a webcam, it can be
+ interesting to share the pipeline with multiple clients. This must be done when
+ only one instance of the video capture element can be used at a time. In this
+ case, the shared property of GstRTSPMedia must be used to instruct the default
+ GstRTSPMediaFactory implementation to cache the media.
+
+ When all objects created from a factory can be shared, you can set the shared
+ property directly on the factory.
+
+* more on GstRTSPMedia
+
+ After creating the GstRTSPMedia object from the factory, it can be prepared
+ with gst_rtsp_media_prepare(). This method will put those objects in a
+ GstPipeline and will construct and link the streaming elements and the
+ rtpbin session manager object.
+
+ The _prepare() method will then preroll the pipeline in order to figure out the
+ caps on the payloaders. After the GstRTSPMedia prerolled it will be in the
+ prepared state and can be used for creating SDP files or for streaming to
+ clients.
+
+ The prepare method will also create 2 UDP ports for each stream that can be
+ used for sending and receiving RTP/RTCP from clients. These port numbers will
+ have to be negotiated with the client in the SETUP requests.
+
+ When preparing a GstRTSPMedia, an appsink and asppsrc is also constructed
+ for streaming the stream over TCP when requested.
+
+ Media is prepared by the server when DESCRIBE or SETUP requests are received
+ from the client.
+
+
+* the GstRTSPClient object
+
+ When a server detects a new client connection on its port, it will accept the
+ connection, check if the connection is allowed and then call the vmethod
+ create_client. The default implementation of this function will create
+ a new GstRTCPClient object, will configure the session pool, mount points,
+ auth and thread pool objects in it.
+
+ The server will then attach the new client to a server mainloop to let it
+ handle further communication with the client. In RTSP it is usual to keep
+ the connection open between multiple RTSP requests. The client watch will
+ be dispatched by the server mainloop when a new GstRTSPMessage is received,
+ which will then be handled and a response will be sent.
+
+ The GstRTSPClient object remains alive for as long as a client has a TCP
+ connection open with the server. Since is possible for a client to open and close
+ the TCP connection between requests, we cannot store the state related
+ to the configured RTSP session in the GstRTSPClient object. This server state
+ is instead stored in the GstRTSPSession object, identified with the session
+ id.
+
+
+* GstRTSPSession
+
+ This object contains state about a specific RTSP session identified with a
+ session id. This state contains the configured streams and their associated
+ transports.
+
+ When a GstRTSPClient performs a SETUP request, the server will allocate a new
+ GstRTSPSession with a unique session id from the GstRTSPSessionPool. The pool
+ maintains a list of all existing sessions and makes sure that no session id is
+ used multiple times. The session id is sent to the client so that the client
+ can refer to its previously configured state by sending the session id in
+ further requests.
+
+ A client will then use the session id to configure one or more
+ GstRTSPSessionMedia objects, identified by their url. This SessionMedia object
+ contains the configuration of a GstRTSPMedia and its configured
+ GstRTSPStreamTransport.
+
+
+* GstRTSPSessionMedia and GstRTSPStreamTransport
+
+ A GstRTSPSessionMedia is identified by a URL and is referenced by a
+ GstRTSPSession. It is created as soon as a client performs a SETUP operation on
+ a particular URL. It will contain a link to the GstRTSPMedia object associated
+ with the URL along with the state of the media and the configured transports
+ for each of the streams in the media.
+
+ Each SETUP request performed by the client will configure a
+ GstRTSPStreamTransport object linked to by the GstRTSPSessionMedia structure.
+ It will contain the transport information needed to send this stream to the
+ client. The GstRTSPStreamTransport also contains a link to the GstRTSPStream
+ object that generates the actual data to be streamed to the client.
+
+ Note how GstRTSPMedia and GstRTSPStream (the providers of the data to
+ stream) are decoupled from GstRTSPSessionMedia and GstRTSPStreamTransport (the
+ configuration of how to send this stream to a client) in order to be able to
+ send the data of one GstRTSPMedia to multiple clients.
+
+
+* media control
+
+ After a client has configured the transports for a GstRTSPMedia and its
+ GstRTSPStreams, the client can play/pause/stop the stream.
+
+ The GstRTSPMedia object was prepared in the DESCRIBE call (or during SETUP when
+ the client skipped the DESCRIBE request). As seen earlier, this configures a
+ couple of udpsink and udpsrc elements to respectively send and receive the
+ media to clients.
+
+ When a client performs a PLAY request, its configured destination UDP ports are
+ added to the GstRTSPStream target destinations, at which point data will
+ be sent to the client. The corresponding GstRTSPMedia object will be set to the
+ PLAYING state if it was not allready in order to send the data to the
+ destination.
+
+ The server needs to prepare an RTP-Info header field in the PLAY response,
+ which consists of the sequence number and the RTP timestamp of the next RTP
+ packet. In order to achive this, the server queries the payloaders for this
+ information when it prerolled the pipeline.
+
+ When a client performs a PAUSE request, the destination UDP ports are removed
+ from the GstRTSPStream object and the GstRTSPMedia object is set to PAUSED
+ if no other destinations are configured anymore.
+
+
+* seeking
+
+ A seek is performed when a client sends a Range header in the PLAY request.
+ This only works when not dealing with shared (live) streams.
+
+ The server performs a GStreamer flushing seek on the media, waits for the
+ pipeline to preroll again and then responds to the client after collecting the
+ new RTP sequence number and timestamp from the payloaders.
+
+
+* session management
+
+ The server has to react to clients that suddenly disappear because of network
+ problems or otherwise. It needs to make sure that it can reasonable free the
+ resources that are used by the various objects in use for streaming when the
+ client appears to be gone.
+
+ Each of the GstRTSPSession objects managed by a GstRTSPSessionPool has
+ therefore a last_access field that contains the timestamp of when activity from
+ a client was last recorded.
+
+ Various ways exist to detect activity from a client:
+
+ - RTSP keepalive requests. When a client is receiving RTP data, the RTSP TCP
+ connection is largely unused. It is the client's responsability to
+ periodically send keep-alive requests over the TCP channel.
+
+ Whenever a keep-alive request is received by the server (any request that
+ contains a session id, usually an OPTION or GET_PARAMETER request) the
+ last_access of the session is updated.
+
+ - Since it is not required for a client to keep the RTSP TCP connection open
+ while streaming, gst-rtsp-server also detects activity from clients by
+ looking at the RTCP messages it receives.
+
+ When an RTCP message is received from a client, the server looks in its list
+ of active ports if this message originates from a known host/port pair that
+ is currently active in a GstRTSPSession. If this is the case, the session is
+ kept alive.
+
+ Since the server does not know anything about the port number that will be
+ used by the client to send RTCP, this method does not always work. Later
+ RTSP RFCs will include support for negotiating this port number with the
+ server. Most clients however use the same port number for sending and
+ receiving RTCP exactly for this reason.
+
+ If there was no activity in a particular session for a long time (by default 60
+ seconds), the application should remove the session from the pool. For this,
+ the application should periodically (say every 2 seconds) check if no sessions
+ expired and call gst_rtsp_session_pool_cleanup() to remove them.
+
+ When a session is removed from the sessionpool and its last reference is
+ unreffef, all related objects and media are destroyed as if a TEARDOWN happened
+ from the client.
+
+
+* TEARDOWN
+
+ A TEARDOWN request will first locate the GstRTSPSessionMedia of the URL. It
+ will then remove all transports from the streams, making sure that streaming
+ stops to the clients. It will then remove the GstRTSPSessionMedia and
+ GstRTSPStreamTransport objects. Finally the GstRTSPSession is released back
+ into the pool.
+
+ When there are no more references to the GstRTSPMedia, the media pipeline is
+ shut down (with _unprepare) and destroyed. This will then also destroy the
+ GstRTSPStream objects.
+
+
+* Security
+
+ The security of the server and the policy is implemented in a GstRTSPAuth
+ object. The object is reponsible for:
+
+ - authenticate the user of the server.
+
+ - check if the current user is authorized to perform an operation.
+
+ For critical operations, the server will call gst_rtsp_auth_check() with
+ a string describing the operation which should be validated. The installed
+ GstRTSPAuth object is then responsible for checking if the operation is
+ allowed.
+
+ Implementations of GstRTSPAuth objects can use the following infrastructure
+ bits of the rtsp server to implement these checks:
+
+ - GstRTSPToken: a generic structure describing roles and permissions granted
+ to a user.
+
+ - GstRTSPPermissions: a generic list of roles and matching permissions. These
+ can be attached to media and facties currently.
+
+ An Auth implementation will usually authenticate a user, using method such as
+ Basic authentication or client certificates or perhaps simply use the IP address.
+ The result of the authentication of the user will be a GstRTSPToken that is made
+ current in the context of the ongoing request.
+
+ The auth module can then implement the various checks in the server by looking
+ at the current token and, if needed, compare it to the required GstRTSPPermissions
+ of the current object.
+
+ The security is deliberately kept generic with a default implementation of the
+ GstRTSPAuth object providing a usable and simple implementaion. To make more
+ complicated security modules, the auth object should be subclassed and new
+ implementations for the checks needs to be made.
+
+
+Objects
+-------
+
+GstRTSPServer
+ - Toplevel object listening for connections and creating new
+ GstRTSPClient objects
+
+GstRTSPClient
+ - Handle RTSP Requests from connected clients. All other objects
+ are called by this object.
+
+GstRTSPContext
+ - Helper structure contaning the current state of the request
+ handled by the client.
+
+
+GstRTSPMountPoints
+ - Maps a url to a GstRTSPMediaFactory implementation. The default
+ implementation uses a simple hashtable to map a url to a factory.
+
+GstRTSPMediaFactory
+ - Creates and caches GstRTSPMedia objects. The default implementation
+ can create GstRTSPMedia objects based on gst-launch syntax.
+
+GstRTSPMediaFactoryURI
+ - Specialized GstRTSPMediaFactory that can stream the content of any
+ URI.
+
+GstRTSPMedia
+ - The object that contains the media pipeline and various GstRTSPStream
+ objects that produce RTP packets
+
+GstRTSPStream
+ - Manages the elements to stream a stream of a GstRTSPMedia to one or
+ more GstRTSPStreamTransports.
+
+
+GstRTSPSessionPool
+ - Creates and manages GstRTSPSession objects identified by an id.
+
+GstRTSPSession
+ - An object containing the various GstRTSPSessionMedia objects managed
+ by this session.
+
+GstRTSPSessionMedia
+ - The state of a GstRTSPMedia and the configuration of a GstRTSPStream
+ objects. The configuration for the GstRTSPStream is stored in
+ GstRTSPStreamTransport objects.
+
+GstRTSPStreamTransport
+ - Configuration of how a GstRTSPStream is send to a particular client. It
+ contains the transport that was negotiated with the client in the SETUP
+ request.
+
+
+GstRTSPSDP
+ - helper functions for creating SDP messages from gstRTSPMedia
+
+GstRTSPAddressPool
+ - a pool of multicast and unicast addresses used in streaming
+
+GstRTSPThreadPool
+ - a pool of threads used for various server tasks such as handling clients and
+ managin media pipelines.
+
+
+GstRTSPAuth
+ - Hooks for checking authorizations, all client activity will call this
+ object with the GstRTSPContext structure. By default it supports
+ basic authentication.
+
+GstRTSPToken
+ - Credentials of a user. This contrains the roles that the user is allowed
+ to assume and other permissions or capabilities of the user.
+
+GstRTSPPermissions
+ - A list of permissions for each role. The permissions are usually attached
+ to objects to describe what roles have what permissions.
+
+GstRTSPParams
+ - object to handle get and set parameter requests.
+
--- /dev/null
+RTSP server
+-----------
+
+This directory contains an example RTSP server built with various GStreamer
+components and libraries. It also uses GStreamer for all of the multimedia
+procesing and RTP bits. The following features are implemented:
+
+ -
+
+Server Design
+-------------
+
+The toplevel component of the server is a GstRTSPServer object. This object
+creates and binds on the server socket and attaches into the mainloop.
+
+For each request a new GstRTSPClient object is created that will accept the
+request and a thread is started to handle further communication with the
+client until the connection is closed.
+
+When a client issues a SETUP request we create a GstRTSPSession object,
+identified with a sessionid, that will keep track of the state of a client.
+The object is destroyed when a TEARDOWN request is made for that sessionid.
+
+We also maintain a pool of URL to media pipeline mappings. Each url is mapped to
+an object that is able to provide a pipeline for that media. We provide
+pipelines to stream live captured data, on-demand file streaming or on-demand
+transcoding of a file or stream.
+
+A pool of currently active pipelines is also maintained. Usually the active
+pipelines are in use by one or more GstRTSPSession objects. An active pipeline
+becomes inactive when no more sessions refer to it.
+
+A client can choose to start a new pipeline or join a currently active pipeline.
+Some active pipeline cannot be joined (such as on-demand streams) but a new
+instance of that pipeline can be created.
--- /dev/null
+GST_DOC_SCANOBJ = $(top_srcdir)/common/gstdoc-scangobj
+
+## Process this file with automake to produce Makefile.in
+
+# The name of the module, e.g. 'glib'.
+MODULE=gst-rtsp-server
+DOC_MODULE=$(MODULE)
+
+# for upload-doc.mak
+DOC=$(MODULE)
+FORMATS=html
+html: html-build.stamp
+include $(top_srcdir)/common/upload-doc.mak
+
+# generated basefiles
+#basefiles = \
+## $(DOC_MODULE).types \
+# $(DOC_MODULE)-sections.txt \
+# $(DOC_MODULE)-docs.sgml
+
+# ugly hack to make -unused.sgml work
+#unused-build.stamp:
+# BUILDDIR=`pwd` && \
+# cd $(srcdir)/tmpl && \
+# ln -sf gstreamer-libs-unused.sgml \
+# $$BUILDDIR/tmpl/gstreamer-libs-@GST_API_VERSION@-unused.sgml
+# touch unused-build.stamp
+
+# these rules are added to create parallel docs using GST_API_VERSION
+#$(basefiles): gstreamer-libs-@GST_API_VERSION@%: gstreamer-libs%
+# cp $< $@
+
+#CLEANFILES = $(basefiles)
+
+# The top-level SGML file. Change it if you want.
+DOC_MAIN_SGML_FILE=$(DOC_MODULE)-docs.sgml
+
+# The directory containing the source code. Relative to $(top_srcdir).
+# gtk-doc will search all .c & .h files beneath here for inline comments
+# documenting functions and macros.
+DOC_SOURCE_DIR = $(top_srcdir)/gst/rtsp-server/
+DOC_BUILD_DIR=$(top_builddir)/gst/rtsp-server/
+
+SCAN_OPTIONS=
+
+# FIXME :
+# there's something wrong with gstreamer-sections.txt not being in the dist
+# maybe it doesn't resolve; we're adding it below for now
+#EXTRA_DIST = gstreamer.types.in gstreamer.hierarchy $(DOC_MODULE)-sections.txt gstreamer-sections.txt $(DOC_MAIN_SGML_FILE)
+
+# Extra options to supply to gtkdoc-mkdb.
+MKDB_OPTIONS=--sgml-mode --source-suffixes=c,h,cc,m
+
+# Extra options to supply to gtkdoc-fixref.
+FIXXREF_OPTIONS=--extra-dir=$(GLIB_PREFIX)/share/gtk-doc/html \
+ --extra-dir=$(GST_PREFIX)/share/gtk-doc/html \
+ --extra-dir=$(GSTPB_PREFIX)/share/gtk-doc/html
+
+# Used for dependencies.
+HFILE_GLOB=$(DOC_SOURCE_DIR)/*.h
+CFILE_GLOB=$(DOC_SOURCE_DIR)/*.c
+
+SCANOBJ_DEPS = \
+ $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la
+
+# Extra options to supply to gtkdoc-scan.
+SCANOBJ_OPTIONS=--type-init-func="g_type_init();gst_init(&argc,&argv)"
+
+# Header files to ignore when scanning.
+IGNORE_HFILES =
+IGNORE_CFILES =
+
+# we add all .h files of elements that have signals/args we want
+# sadly this also pulls in the private methods - maybe we should
+# move those around in the source ?
+# also, we should add some stuff here conditionally based on whether
+# or not the plugin will actually build
+# but I'm not sure about that - it might be this Just Works given that
+# the registry won't have the element
+
+EXTRA_HFILES =
+
+# Images to copy into HTML directory.
+HTML_IMAGES =
+
+# Extra SGML files that are included by $(DOC_MAIN_SGML_FILE).
+content_files =
+
+# Other files to distribute.
+extra_files =
+
+# CFLAGS and LDFLAGS for compiling scan program. Only needed if your app/lib
+# contains GtkObjects/GObjects and you want to document signals and properties.
+GTKDOC_CFLAGS = -I$(top_srcdir) $(GST_PLUGINS_BASE_CFLAGS) \
+ $(GST_BASE_CFLAGS) $(GST_CFLAGS)
+GTKDOC_LIBS = $(SCANOBJ_DEPS) $(GST_PLUGINS_BASE_LIBS) \
+ $(GST_BASE_LIBS) $(GST_LIBS) $(GCOV_LIBS)
+
+GTKDOC_CC=$(LIBTOOL) --tag=CC --mode=compile $(CC)
+GTKDOC_LD=$(LIBTOOL) --tag=CC --mode=link $(CC)
+
+# If you need to override some of the declarations, place them in this file
+# and uncomment this line.
+DOC_OVERRIDES = $(DOC_MODULE)-overrides.txt
+
+include $(top_srcdir)/common/gtk-doc.mak
--- /dev/null
+# Makefile.in generated by automake 1.14.1 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994-2013 Free Software Foundation, Inc.
+
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+
+# this snippet is to be included by both our docbook manuals
+# and gtk-doc API references
+
+# it adds an upload target to each of these dir's Makefiles
+
+# each Makefile.am should define the following variables:
+# - DOC: the base name of the documentation
+# (faq, manual, pwg, gstreamer, gstreamer-libs)
+# - FORMATS: the formats in which DOC is output
+# (html ps pdf)
+
+# if you want to use it, make sure your $HOME/.ssh/config file contains the
+# correct User entry for the Host entry for the DOC_SERVER
+
+###########################################################################
+# Everything below here is generic and you shouldn't need to change it.
+###########################################################################
+# thomas: except of course that we did
+VPATH = @srcdir@
+am__is_gnu_make = test -n '$(MAKEFILE_LIST)' && test -n '$(MAKELEVEL)'
+am__make_running_with_option = \
+ case $${target_option-} in \
+ ?) ;; \
+ *) echo "am__make_running_with_option: internal error: invalid" \
+ "target option '$${target_option-}' specified" >&2; \
+ exit 1;; \
+ esac; \
+ has_opt=no; \
+ sane_makeflags=$$MAKEFLAGS; \
+ if $(am__is_gnu_make); then \
+ sane_makeflags=$$MFLAGS; \
+ else \
+ case $$MAKEFLAGS in \
+ *\\[\ \ ]*) \
+ bs=\\; \
+ sane_makeflags=`printf '%s\n' "$$MAKEFLAGS" \
+ | sed "s/$$bs$$bs[$$bs $$bs ]*//g"`;; \
+ esac; \
+ fi; \
+ skip_next=no; \
+ strip_trailopt () \
+ { \
+ flg=`printf '%s\n' "$$flg" | sed "s/$$1.*$$//"`; \
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+ for flg in $$sane_makeflags; do \
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+transform = $(program_transform_name)
+NORMAL_INSTALL = :
+PRE_INSTALL = :
+POST_INSTALL = :
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+host_triplet = @host@
+target_triplet = @target@
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+DEFS = @DEFS@
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+GCOV_LIBS = @GCOV_LIBS@
+GIO_CFLAGS = @GIO_CFLAGS@
+GIO_LDFLAGS = @GIO_LDFLAGS@
+GIO_LIBS = @GIO_LIBS@
+GLIB_CFLAGS = @GLIB_CFLAGS@
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+GLIB_GENMARSHAL = @GLIB_GENMARSHAL@
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+GLIB_MKENUMS = @GLIB_MKENUMS@
+GLIB_REQ = @GLIB_REQ@
+GREP = @GREP@
+GSTPB_PLUGINS_DIR = @GSTPB_PLUGINS_DIR@
+GSTPD_PLUGINS_DIR = @GSTPD_PLUGINS_DIR@
+GSTPG_PLUGINS_DIR = @GSTPG_PLUGINS_DIR@
+GST_AGE = @GST_AGE@
+GST_ALL_CFLAGS = @GST_ALL_CFLAGS@
+GST_ALL_LDFLAGS = @GST_ALL_LDFLAGS@
+GST_ALL_LIBS = @GST_ALL_LIBS@
+GST_API_VERSION = @GST_API_VERSION@
+GST_BASE_CFLAGS = @GST_BASE_CFLAGS@
+GST_BASE_LIBS = @GST_BASE_LIBS@
+GST_CFLAGS = @GST_CFLAGS@
+GST_CHECK_CFLAGS = @GST_CHECK_CFLAGS@
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+GST_PLUGINS_BASE_CFLAGS = @GST_PLUGINS_BASE_CFLAGS@
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+GTKDOC_DEPS_LIBS = @GTKDOC_DEPS_LIBS@
+GTKDOC_MKPDF = @GTKDOC_MKPDF@
+GTKDOC_REBASE = @GTKDOC_REBASE@
+HAVE_DOCBOOK2PS = @HAVE_DOCBOOK2PS@
+HAVE_DVIPS = @HAVE_DVIPS@
+HAVE_EPSTOPDF = @HAVE_EPSTOPDF@
+HAVE_JADETEX = @HAVE_JADETEX@
+HAVE_PNGTOPNM = @HAVE_PNGTOPNM@
+HAVE_PNMTOPS = @HAVE_PNMTOPS@
+HAVE_PS2PDF = @HAVE_PS2PDF@
+HAVE_XMLLINT = @HAVE_XMLLINT@
+HAVE_XSLTPROC = @HAVE_XSLTPROC@
+HTML_DIR = @HTML_DIR@
+INSTALL = @INSTALL@
+INSTALL_DATA = @INSTALL_DATA@
+INSTALL_PROGRAM = @INSTALL_PROGRAM@
+INSTALL_SCRIPT = @INSTALL_SCRIPT@
+INSTALL_STRIP_PROGRAM = @INSTALL_STRIP_PROGRAM@
+INTROSPECTION_CFLAGS = @INTROSPECTION_CFLAGS@
+INTROSPECTION_COMPILER = @INTROSPECTION_COMPILER@
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+INTROSPECTION_MAKEFILE = @INTROSPECTION_MAKEFILE@
+INTROSPECTION_SCANNER = @INTROSPECTION_SCANNER@
+INTROSPECTION_TYPELIBDIR = @INTROSPECTION_TYPELIBDIR@
+LD = @LD@
+LDFLAGS = @LDFLAGS@
+LIBCGROUP_CFLAGS = @LIBCGROUP_CFLAGS@
+LIBCGROUP_LIBS = @LIBCGROUP_LIBS@
+LIBOBJS = @LIBOBJS@
+LIBS = @LIBS@
+LIBTOOL = @LIBTOOL@
+LIPO = @LIPO@
+LN_S = @LN_S@
+LTLIBOBJS = @LTLIBOBJS@
+MAINT = @MAINT@
+MAKEINFO = @MAKEINFO@
+MANIFEST_TOOL = @MANIFEST_TOOL@
+MKDIR_P = @MKDIR_P@
+NM = @NM@
+NMEDIT = @NMEDIT@
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+PACKAGE_VERSION_NANO = @PACKAGE_VERSION_NANO@
+PACKAGE_VERSION_RELEASE = @PACKAGE_VERSION_RELEASE@
+PATH_SEPARATOR = @PATH_SEPARATOR@
+PKG_CONFIG = @PKG_CONFIG@
+PLUGINDIR = @PLUGINDIR@
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+RANLIB = @RANLIB@
+SED = @SED@
+SET_MAKE = @SET_MAKE@
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+VERSION = @VERSION@
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+mandir = @mandir@
+mkdir_p = @mkdir_p@
+oldincludedir = @oldincludedir@
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+prefix = @prefix@
+program_transform_name = @program_transform_name@
+psdir = @psdir@
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+top_srcdir = @top_srcdir@
+GST_DOC_SCANOBJ = $(top_srcdir)/common/gstdoc-scangobj
+
+# The name of the module, e.g. 'glib'.
+MODULE = gst-rtsp-server
+DOC_MODULE = $(MODULE)
+
+# for upload-doc.mak
+DOC = $(MODULE)
+FORMATS = html
+
+# these variables define the location of the online docs
+DOC_SERVER = gstreamer.freedesktop.org
+DOC_BASE = /srv/gstreamer.freedesktop.org/www/data/doc
+DOC_URL = $(DOC_SERVER):$(DOC_BASE)
+
+# generated basefiles
+#basefiles = \
+# $(DOC_MODULE)-sections.txt \
+# $(DOC_MODULE)-docs.sgml
+
+# ugly hack to make -unused.sgml work
+#unused-build.stamp:
+# BUILDDIR=`pwd` && \
+# cd $(srcdir)/tmpl && \
+# ln -sf gstreamer-libs-unused.sgml \
+# $$BUILDDIR/tmpl/gstreamer-libs-@GST_API_VERSION@-unused.sgml
+# touch unused-build.stamp
+
+# these rules are added to create parallel docs using GST_API_VERSION
+#$(basefiles): gstreamer-libs-@GST_API_VERSION@%: gstreamer-libs%
+# cp $< $@
+
+#CLEANFILES = $(basefiles)
+
+# The top-level SGML file. Change it if you want.
+DOC_MAIN_SGML_FILE = $(DOC_MODULE)-docs.sgml
+
+# The directory containing the source code. Relative to $(top_srcdir).
+# gtk-doc will search all .c & .h files beneath here for inline comments
+# documenting functions and macros.
+DOC_SOURCE_DIR = $(top_srcdir)/gst/rtsp-server/
+DOC_BUILD_DIR = $(top_builddir)/gst/rtsp-server/
+SCAN_OPTIONS =
+
+# FIXME :
+# there's something wrong with gstreamer-sections.txt not being in the dist
+# maybe it doesn't resolve; we're adding it below for now
+#EXTRA_DIST = gstreamer.types.in gstreamer.hierarchy $(DOC_MODULE)-sections.txt gstreamer-sections.txt $(DOC_MAIN_SGML_FILE)
+
+# Extra options to supply to gtkdoc-mkdb.
+MKDB_OPTIONS = --sgml-mode --source-suffixes=c,h,cc,m
+
+# Extra options to supply to gtkdoc-fixref.
+FIXXREF_OPTIONS = --extra-dir=$(GLIB_PREFIX)/share/gtk-doc/html \
+ --extra-dir=$(GST_PREFIX)/share/gtk-doc/html \
+ --extra-dir=$(GSTPB_PREFIX)/share/gtk-doc/html
+
+
+# Used for dependencies.
+HFILE_GLOB = $(DOC_SOURCE_DIR)/*.h
+CFILE_GLOB = $(DOC_SOURCE_DIR)/*.c
+SCANOBJ_DEPS = \
+ $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la
+
+
+# Extra options to supply to gtkdoc-scan.
+SCANOBJ_OPTIONS = --type-init-func="g_type_init();gst_init(&argc,&argv)"
+
+# Header files to ignore when scanning.
+IGNORE_HFILES =
+IGNORE_CFILES =
+
+# we add all .h files of elements that have signals/args we want
+# sadly this also pulls in the private methods - maybe we should
+# move those around in the source ?
+# also, we should add some stuff here conditionally based on whether
+# or not the plugin will actually build
+# but I'm not sure about that - it might be this Just Works given that
+# the registry won't have the element
+EXTRA_HFILES =
+
+# Images to copy into HTML directory.
+HTML_IMAGES =
+
+# Extra SGML files that are included by $(DOC_MAIN_SGML_FILE).
+content_files =
+
+# Other files to distribute.
+extra_files =
+
+# CFLAGS and LDFLAGS for compiling scan program. Only needed if your app/lib
+# contains GtkObjects/GObjects and you want to document signals and properties.
+GTKDOC_CFLAGS = -I$(top_srcdir) $(GST_PLUGINS_BASE_CFLAGS) \
+ $(GST_BASE_CFLAGS) $(GST_CFLAGS)
+
+GTKDOC_LIBS = $(SCANOBJ_DEPS) $(GST_PLUGINS_BASE_LIBS) \
+ $(GST_BASE_LIBS) $(GST_LIBS) $(GCOV_LIBS)
+
+GTKDOC_CC = $(LIBTOOL) --tag=CC --mode=compile $(CC)
+GTKDOC_LD = $(LIBTOOL) --tag=CC --mode=link $(CC)
+
+# If you need to override some of the declarations, place them in this file
+# and uncomment this line.
+DOC_OVERRIDES = $(DOC_MODULE)-overrides.txt
+
+# thomas: copied from glib-2
+# We set GPATH here; this gives us semantics for GNU make
+# which are more like other make's VPATH, when it comes to
+# whether a source that is a target of one rule is then
+# searched for in VPATH/GPATH.
+#
+GPATH = $(srcdir)
+
+# thomas: make docs parallel installable
+TARGET_DIR = $(HTML_DIR)/$(DOC_MODULE)-@GST_API_VERSION@
+EXTRA_DIST = \
+ $(content_files) \
+ $(extra_files) \
+ $(HTML_IMAGES) \
+ $(DOC_MAIN_SGML_FILE) \
+ $(DOC_MODULE).types \
+ $(DOC_OVERRIDES) \
+ $(DOC_MODULE)-sections.txt
+
+DOC_STAMPS = \
+ setup-build.stamp \
+ scan-build.stamp \
+ sgml-build.stamp \
+ html-build.stamp \
+ sgml.stamp \
+ html.stamp
+
+SCANOBJ_FILES = \
+ $(DOC_MODULE).args \
+ $(DOC_MODULE).hierarchy \
+ $(DOC_MODULE).interfaces \
+ $(DOC_MODULE).prerequisites \
+ $(DOC_MODULE).signals \
+ .libs/$(DOC_MODULE)-scan.o
+
+REPORT_FILES = \
+ $(DOC_MODULE)-undocumented.txt \
+ $(DOC_MODULE)-undeclared.txt \
+ $(DOC_MODULE)-unused.txt
+
+CLEANFILES = $(SCANOBJ_FILES) $(REPORT_FILES) $(DOC_STAMPS) doc-registry.xml
+all: all-am
+
+.SUFFIXES:
+$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ $(srcdir)/Makefile.am $(top_srcdir)/common/upload-doc.mak $(top_srcdir)/common/gtk-doc.mak $(am__configure_deps)
+ @for dep in $?; do \
+ case '$(am__configure_deps)' in \
+ *$$dep*) \
+ ( cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh ) \
+ && { if test -f $@; then exit 0; else break; fi; }; \
+ exit 1;; \
+ esac; \
+ done; \
+ echo ' cd $(top_srcdir) && $(AUTOMAKE) --gnu docs/libs/Makefile'; \
+ $(am__cd) $(top_srcdir) && \
+ $(AUTOMAKE) --gnu docs/libs/Makefile
+.PRECIOUS: Makefile
+Makefile: $(srcdir)/Makefile.in $(top_builddir)/config.status
+ @case '$?' in \
+ *config.status*) \
+ cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh;; \
+ *) \
+ echo ' cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe)'; \
+ cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe);; \
+ esac;
+$(top_srcdir)/common/upload-doc.mak $(top_srcdir)/common/gtk-doc.mak:
+
+$(top_builddir)/config.status: $(top_srcdir)/configure $(CONFIG_STATUS_DEPENDENCIES)
+ cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+
+$(top_srcdir)/configure: @MAINTAINER_MODE_TRUE@ $(am__configure_deps)
+ cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(ACLOCAL_M4): @MAINTAINER_MODE_TRUE@ $(am__aclocal_m4_deps)
+ cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(am__aclocal_m4_deps):
+
+mostlyclean-libtool:
+ -rm -f *.lo
+
+clean-libtool:
+ -rm -rf .libs _libs
+tags TAGS:
+
+ctags CTAGS:
+
+cscope cscopelist:
+
+
+distdir: $(DISTFILES)
+ @srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+ topsrcdirstrip=`echo "$(top_srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+ list='$(DISTFILES)'; \
+ dist_files=`for file in $$list; do echo $$file; done | \
+ sed -e "s|^$$srcdirstrip/||;t" \
+ -e "s|^$$topsrcdirstrip/|$(top_builddir)/|;t"`; \
+ case $$dist_files in \
+ */*) $(MKDIR_P) `echo "$$dist_files" | \
+ sed '/\//!d;s|^|$(distdir)/|;s,/[^/]*$$,,' | \
+ sort -u` ;; \
+ esac; \
+ for file in $$dist_files; do \
+ if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+ if test -d $$d/$$file; then \
+ dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+ if test -d "$(distdir)/$$file"; then \
+ find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+ fi; \
+ if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+ cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+ find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+ fi; \
+ cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+ else \
+ test -f "$(distdir)/$$file" \
+ || cp -p $$d/$$file "$(distdir)/$$file" \
+ || exit 1; \
+ fi; \
+ done
+ $(MAKE) $(AM_MAKEFLAGS) \
+ top_distdir="$(top_distdir)" distdir="$(distdir)" \
+ dist-hook
+check-am: all-am
+check: check-am
+all-am: Makefile all-local
+installdirs:
+install: install-am
+install-exec: install-exec-am
+install-data: install-data-am
+uninstall: uninstall-am
+
+install-am: all-am
+ @$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-am
+install-strip:
+ if test -z '$(STRIP)'; then \
+ $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+ install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+ install; \
+ else \
+ $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+ install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+ "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+ fi
+mostlyclean-generic:
+
+clean-generic:
+ -test -z "$(CLEANFILES)" || rm -f $(CLEANFILES)
+
+distclean-generic:
+ -test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+ -test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+ @echo "This command is intended for maintainers to use"
+ @echo "it deletes files that may require special tools to rebuild."
+clean: clean-am
+
+clean-am: clean-generic clean-libtool clean-local mostlyclean-am
+
+distclean: distclean-am
+ -rm -f Makefile
+distclean-am: clean-am distclean-generic distclean-local
+
+dvi: dvi-am
+
+dvi-am:
+
+html-am:
+
+info: info-am
+
+info-am:
+
+install-data-am: install-data-local
+
+install-dvi: install-dvi-am
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-am
+
+install-html-am:
+
+install-info: install-info-am
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-am
+
+install-pdf-am:
+
+install-ps: install-ps-am
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-am
+ -rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic \
+ maintainer-clean-local
+
+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-generic mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
+
+ps: ps-am
+
+ps-am:
+
+uninstall-am: uninstall-local
+
+.MAKE: install-am install-strip
+
+.PHONY: all all-am all-local check check-am clean clean-generic \
+ clean-libtool clean-local cscopelist-am ctags-am dist-hook \
+ distclean distclean-generic distclean-libtool distclean-local \
+ distdir dvi dvi-am html html-am info info-am install \
+ install-am install-data install-data-am install-data-local \
+ install-dvi install-dvi-am install-exec install-exec-am \
+ install-html install-html-am install-info install-info-am \
+ install-man install-pdf install-pdf-am install-ps \
+ install-ps-am install-strip installcheck installcheck-am \
+ installdirs maintainer-clean maintainer-clean-generic \
+ maintainer-clean-local mostlyclean mostlyclean-generic \
+ mostlyclean-libtool pdf pdf-am ps ps-am tags-am uninstall \
+ uninstall-am uninstall-local
+
+html: html-build.stamp
+
+upload: $(FORMATS)
+ @if echo $(FORMATS) | grep html > /dev/null; then \
+ echo "Preparing docs for upload (rebasing cross-references) ..." ; \
+ if test x$(builddir) != x$(srcdir); then \
+ echo "make upload can only be used if srcdir == builddir"; \
+ exit 1; \
+ fi; \
+ # gtkdoc-rebase sometimes gets confused, so reset everything to \
+ # local links before rebasing to online links \
+ gtkdoc-rebase --html-dir=$(builddir)/html 2>/dev/null 2>/dev/null ; \
+ rebase=`gtkdoc-rebase --verbose --online --html-dir=$(builddir)/html` ; \
+ echo "$$rebase" | grep -e "On-*line"; \
+ for req in glib gobject gstreamer gstreamer-libs gst-plugins-base-libs; do \
+ if ! ( echo "$$rebase" | grep -i -e "On-*line.*/$$req/" ); then \
+ echo "===============================================================================" ; \
+ echo " Could not determine online location for $$req docs. Cross-referencing will be " ; \
+ echo " broken, so not uploading. Make sure the library's gtk-doc documentation is " ; \
+ echo " installed somewhere in /usr/share/gtk-doc. " ; \
+ echo "===============================================================================" ; \
+ exit 1; \
+ fi; \
+ done; \
+ export SRC="$$SRC html"; \
+ fi; \
+ if echo $(FORMATS) | grep ps > /dev/null; then export SRC="$$SRC $(DOC).ps"; fi; \
+ if echo $(FORMATS) | grep pdf > /dev/null; then export SRC="$$SRC $(DOC).pdf"; fi; \
+ \
+ # upload releases to both X.Y/ and head/ subdirectories \
+ export DIR=$(DOC_BASE)/gstreamer/$(PACKAGE_VERSION_MAJOR).$(PACKAGE_VERSION_MINOR)/$(DOC); \
+ echo Uploading $$SRC to $(DOC_SERVER):$$DIR; \
+ ssh $(DOC_SERVER) mkdir -p $$DIR; \
+ rsync -rv -e ssh --delete $$SRC $(DOC_SERVER):$$DIR; \
+ ssh $(DOC_SERVER) chmod -R g+w $$DIR; \
+ \
+ export DIR=$(DOC_BASE)/gstreamer/head/$(DOC); \
+ echo Uploading $$SRC to $(DOC_SERVER):$$DIR; \
+ ssh $(DOC_SERVER) mkdir -p $$DIR; \
+ rsync -rv -e ssh --delete $$SRC $(DOC_SERVER):$$DIR; \
+ ssh $(DOC_SERVER) chmod -R g+w $$DIR; \
+ \
+ if echo $(FORMATS) | grep html > /dev/null; then \
+ echo "Un-preparing docs for upload (rebasing cross-references) ..." ; \
+ gtkdoc-rebase --html-dir=$(builddir)/html ; \
+ fi; \
+ echo Done
+
+@ENABLE_GTK_DOC_TRUE@all-local: html-build.stamp
+
+#### setup ####
+
+@ENABLE_GTK_DOC_TRUE@setup-build.stamp: $(content_files)
+@ENABLE_GTK_DOC_TRUE@ -@if test "$(abs_srcdir)" != "$(abs_builddir)" ; then \
+@ENABLE_GTK_DOC_TRUE@ echo ' DOC Preparing build'; \
+@ENABLE_GTK_DOC_TRUE@ files=`echo $(DOC_MAIN_SGML_FILE) $(DOC_OVERRIDES) $(DOC_MODULE)-sections.txt $(DOC_MODULE).types $(content_files)`; \
+@ENABLE_GTK_DOC_TRUE@ if test "x$$files" != "x" ; then \
+@ENABLE_GTK_DOC_TRUE@ for file in $$files ; do \
+@ENABLE_GTK_DOC_TRUE@ test -f $(abs_srcdir)/$$file && \
+@ENABLE_GTK_DOC_TRUE@ cp -pu $(abs_srcdir)/$$file $(abs_builddir)/ || true; \
+@ENABLE_GTK_DOC_TRUE@ done; \
+@ENABLE_GTK_DOC_TRUE@ fi; \
+@ENABLE_GTK_DOC_TRUE@ fi
+@ENABLE_GTK_DOC_TRUE@ @touch setup-build.stamp
+
+#### scan ####
+
+# in the case of non-srcdir builds, the built gst directory gets added
+# to gtk-doc scanning; but only then, to avoid duplicates
+@ENABLE_GTK_DOC_TRUE@scan-build.stamp: $(HFILE_GLOB) $(CFILE_GLOB)
+@ENABLE_GTK_DOC_TRUE@ @echo ' DOC Scanning header files'
+@ENABLE_GTK_DOC_TRUE@ @_source_dir='' ; \
+@ENABLE_GTK_DOC_TRUE@ for i in $(DOC_SOURCE_DIR) ; do \
+@ENABLE_GTK_DOC_TRUE@ _source_dir="$${_source_dir} --source-dir=$$i" ; \
+@ENABLE_GTK_DOC_TRUE@ done ; \
+@ENABLE_GTK_DOC_TRUE@ gtkdoc-scan \
+@ENABLE_GTK_DOC_TRUE@ $(SCAN_OPTIONS) $(EXTRA_HFILES) \
+@ENABLE_GTK_DOC_TRUE@ --module=$(DOC_MODULE) \
+@ENABLE_GTK_DOC_TRUE@ $${_source_dir} \
+@ENABLE_GTK_DOC_TRUE@ --ignore-headers="$(IGNORE_HFILES)"
+@ENABLE_GTK_DOC_TRUE@ @if grep -l '^..*$$' $(DOC_MODULE).types > /dev/null; then \
+@ENABLE_GTK_DOC_TRUE@ echo " DOC Introspecting gobjects"; \
+@ENABLE_GTK_DOC_TRUE@ GST_PLUGIN_SYSTEM_PATH_1_0=`cd $(top_builddir) && pwd` \
+@ENABLE_GTK_DOC_TRUE@ GST_PLUGIN_PATH_1_0= \
+@ENABLE_GTK_DOC_TRUE@ GST_REGISTRY_1_0=doc-registry.xml \
+@ENABLE_GTK_DOC_TRUE@ $(GTKDOC_EXTRA_ENVIRONMENT) \
+@ENABLE_GTK_DOC_TRUE@ CC="$(GTKDOC_CC)" LD="$(GTKDOC_LD)" \
+@ENABLE_GTK_DOC_TRUE@ CFLAGS="$(GTKDOC_CFLAGS) $(CFLAGS)" \
+@ENABLE_GTK_DOC_TRUE@ LDFLAGS="$(GTKDOC_LIBS) $(LDFLAGS)" \
+@ENABLE_GTK_DOC_TRUE@ gtkdoc-scangobj --type-init-func="gst_init(NULL,NULL)" \
+@ENABLE_GTK_DOC_TRUE@ --module=$(DOC_MODULE) ; \
+@ENABLE_GTK_DOC_TRUE@ else \
+@ENABLE_GTK_DOC_TRUE@ for i in $(SCANOBJ_FILES) ; do \
+@ENABLE_GTK_DOC_TRUE@ test -f $$i || touch $$i ; \
+@ENABLE_GTK_DOC_TRUE@ done \
+@ENABLE_GTK_DOC_TRUE@ fi
+@ENABLE_GTK_DOC_TRUE@ @touch scan-build.stamp
+
+@ENABLE_GTK_DOC_TRUE@$(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(DOC_MODULE)-sections.txt $(DOC_MODULE)-overrides.txt: scan-build.stamp
+@ENABLE_GTK_DOC_TRUE@ @true
+
+#### xml ####
+
+@ENABLE_GTK_DOC_TRUE@sgml-build.stamp: setup-build.stamp $(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(DOC_MODULE)-sections.txt $(expand_content_files)
+@ENABLE_GTK_DOC_TRUE@ @echo ' DOC Building XML'
+@ENABLE_GTK_DOC_TRUE@ @gtkdoc-mkdb --module=$(DOC_MODULE) --source-dir=$(DOC_SOURCE_DIR) --expand-content-files="$(expand_content_files)" --main-sgml-file=$(DOC_MAIN_SGML_FILE) --output-format=xml $(MKDB_OPTIONS)
+@ENABLE_GTK_DOC_TRUE@ @cp ../version.entities xml
+@ENABLE_GTK_DOC_TRUE@ @touch sgml-build.stamp
+
+@ENABLE_GTK_DOC_TRUE@sgml.stamp: sgml-build.stamp
+@ENABLE_GTK_DOC_TRUE@ @true
+
+#### html ####
+
+@ENABLE_GTK_DOC_TRUE@html-build.stamp: sgml.stamp $(DOC_MAIN_SGML_FILE) $(content_files)
+@ENABLE_GTK_DOC_TRUE@ @echo ' DOC Building HTML'
+@ENABLE_GTK_DOC_TRUE@ @rm -rf html
+@ENABLE_GTK_DOC_TRUE@ @mkdir html
+@ENABLE_GTK_DOC_TRUE@ @cp -pr xml html
+@ENABLE_GTK_DOC_TRUE@ @cp ../version.entities ./
+@ENABLE_GTK_DOC_TRUE@ @mkhtml_options=""; \
+@ENABLE_GTK_DOC_TRUE@ gtkdoc-mkhtml 2>&1 --help | grep >/dev/null "\-\-verbose"; \
+@ENABLE_GTK_DOC_TRUE@ if test "$(?)" = "0"; then \
+@ENABLE_GTK_DOC_TRUE@ if test "x$(V)" = "x1"; then \
+@ENABLE_GTK_DOC_TRUE@ mkhtml_options="$$mkhtml_options --verbose"; \
+@ENABLE_GTK_DOC_TRUE@ fi; \
+@ENABLE_GTK_DOC_TRUE@ fi; \
+@ENABLE_GTK_DOC_TRUE@ @gtkdoc-mkhtml 2>&1 --help | grep >/dev/null "\-\-path"; \
+@ENABLE_GTK_DOC_TRUE@ if test "$(?)" = "0"; then \
+@ENABLE_GTK_DOC_TRUE@ mkhtml_options=--path="$(abs_srcdir)"; \
+@ENABLE_GTK_DOC_TRUE@ fi; \
+@ENABLE_GTK_DOC_TRUE@ cd html && gtkdoc-mkhtml $$mkhtml_options $(MKHTML_OPTIONS) $(DOC_MODULE)-@GST_API_VERSION@ ../$(DOC_MAIN_SGML_FILE)
+@ENABLE_GTK_DOC_TRUE@ @rm -rf html/xml
+@ENABLE_GTK_DOC_TRUE@ @rm -f version.entities
+@ENABLE_GTK_DOC_TRUE@ @test "x$(HTML_IMAGES)" = "x" || ( cd $(srcdir) && cp $(HTML_IMAGES) $(abs_builddir)/html )
+@ENABLE_GTK_DOC_TRUE@ @echo ' DOC Fixing cross-references'
+@ENABLE_GTK_DOC_TRUE@ @gtkdoc-fixxref --module=$(DOC_MODULE) --module-dir=html --html-dir=$(HTML_DIR) $(FIXXREF_OPTIONS)
+@ENABLE_GTK_DOC_TRUE@ @touch html-build.stamp
+
+@ENABLE_GTK_DOC_TRUE@clean-local-gtkdoc:
+@ENABLE_GTK_DOC_TRUE@ @rm -rf xml tmpl html
+# clean files copied for nonsrcdir templates build
+@ENABLE_GTK_DOC_TRUE@ @if test x"$(srcdir)" != x. ; then \
+@ENABLE_GTK_DOC_TRUE@ rm -rf $(DOC_MODULE).types; \
+@ENABLE_GTK_DOC_TRUE@ fi
+@ENABLE_GTK_DOC_FALSE@all-local:
+@ENABLE_GTK_DOC_FALSE@clean-local-gtkdoc:
+
+clean-local: clean-local-gtkdoc
+ @rm -f *~ *.bak
+ @rm -rf .libs
+
+distclean-local:
+ @rm -f $(REPORT_FILES) \
+ $(DOC_MODULE)-decl-list.txt $(DOC_MODULE)-decl.txt
+ @rm -rf tmpl/*.sgml.bak
+ @rm -f $(DOC_MODULE).hierarchy
+ @rm -f *.stamp || true
+ @if test "$(abs_srcdir)" != "$(abs_builddir)" ; then \
+ rm -f $(DOC_MAIN_SGML_FILE) ; \
+ rm -f $(DOC_OVERRIDES) ; \
+ rm -f $(DOC_MODULE).types ; \
+ rm -f $(DOC_MODULE).interfaces ; \
+ rm -f $(DOC_MODULE).prerequisites ; \
+ rm -f $(DOC_MODULE)-sections.txt ; \
+ rm -f $(content_files) ; \
+ rm -rf tmpl/*.sgml ; \
+ fi
+ @rm -rf *.o
+
+maintainer-clean-local: clean
+ @cd $(srcdir) && rm -rf html \
+ xml $(DOC_MODULE)-decl-list.txt $(DOC_MODULE)-decl.txt
+
+# thomas: make docs parallel installable; devhelp requires majorminor too
+install-data-local:
+ (installfiles=`echo $(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css`; \
+ if test "$$installfiles" = '$(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css'; \
+ then echo '-- Nothing to install' ; \
+ else \
+ $(mkinstalldirs) $(DESTDIR)$(TARGET_DIR); \
+ for i in $$installfiles; do \
+ echo '-- Installing '$$i ; \
+ $(INSTALL_DATA) $$i $(DESTDIR)$(TARGET_DIR); \
+ done; \
+ echo '-- Installing $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2' ; \
+ if test -e $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; then \
+ $(INSTALL_DATA) $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2 \
+ $(DESTDIR)$(TARGET_DIR)/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; \
+ fi; \
+ $(GTKDOC_REBASE) --relative --dest-dir=$(DESTDIR) --html-dir=$(DESTDIR)$(TARGET_DIR) || true ; \
+ fi)
+uninstall-local:
+ if test -d $(DESTDIR)$(TARGET_DIR); then \
+ rm -rf $(DESTDIR)$(TARGET_DIR)/*; \
+ rmdir -p $(DESTDIR)$(TARGET_DIR) 2>/dev/null || true; \
+ else \
+ echo '-- Nothing to uninstall' ; \
+ fi;
+
+#
+# Require gtk-doc when making dist
+#
+@ENABLE_GTK_DOC_TRUE@dist-check-gtkdoc:
+@ENABLE_GTK_DOC_FALSE@dist-check-gtkdoc:
+@ENABLE_GTK_DOC_FALSE@ @echo "*** gtk-doc must be installed and enabled in order to make dist"
+@ENABLE_GTK_DOC_FALSE@ @false
+
+dist-hook: dist-check-gtkdoc dist-hook-local
+ mkdir $(distdir)/html
+ cp html/* $(distdir)/html
+ -cp $(srcdir)/$(DOC_MODULE).types $(distdir)/
+ -cp $(srcdir)/$(DOC_MODULE)-sections.txt $(distdir)/
+ cd $(distdir) && rm -f $(DISTCLEANFILES)
+ -gtkdoc-rebase --online --relative --html-dir=$(distdir)/html
+
+.PHONY : dist-hook-local docs
+
+# avoid spurious build errors when distchecking with -jN
+.NOTPARALLEL:
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
--- /dev/null
+<?xml version="1.0"?>
+<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook XML V4.1.2//EN"
+ "http://www.oasis-open.org/docbook/xml/4.1.2/docbookx.dtd" [
+<!ENTITY % version-entities SYSTEM "version.entities">
+%version-entities;
+]>
+
+<book id="index" xmlns:xi="http://www.w3.org/2003/XInclude">
+ <bookinfo>
+ <title>GStreamer RTSP Server Reference Manual</title>
+ <releaseinfo>
+ for GStreamer RTSP Server &GST_VERSION;
+ </releaseinfo>
+ </bookinfo>
+
+ <chapter>
+ <xi:include href="xml/rtsp-server.xml"/>
+ <xi:include href="xml/rtsp-client.xml"/>
+ <xi:include href="xml/rtsp-context.xml"/>
+ <xi:include href="xml/rtsp-mount-points.xml"/>
+ <xi:include href="xml/rtsp-media-factory.xml"/>
+ <xi:include href="xml/rtsp-media-factory-uri.xml"/>
+ <xi:include href="xml/rtsp-media.xml"/>
+ <xi:include href="xml/rtsp-stream.xml"/>
+ <xi:include href="xml/rtsp-session-pool.xml"/>
+ <xi:include href="xml/rtsp-session.xml"/>
+ <xi:include href="xml/rtsp-session-media.xml"/>
+ <xi:include href="xml/rtsp-stream-transport.xml"/>
+ <xi:include href="xml/rtsp-sdp.xml"/>
+ <xi:include href="xml/rtsp-address-pool.xml"/>
+ <xi:include href="xml/rtsp-thread-pool.xml"/>
+ <xi:include href="xml/rtsp-auth.xml"/>
+ <xi:include href="xml/rtsp-token.xml"/>
+ <xi:include href="xml/rtsp-permissions.xml"/>
+ <xi:include href="xml/rtsp-params.xml"/>
+ </chapter>
+
+ <chapter id="rtsp-server-hierarchy">
+ <title>Object Hierarchy</title>
+ <xi:include href="xml/tree_index.sgml"/>
+ </chapter>
+
+ <index id="api-index-full">
+ <title>API Index</title>
+ <xi:include href="xml/api-index-full.xml"><xi:fallback /></xi:include>
+ </index>
+
+ <xi:include href="xml/annotation-glossary.xml"><xi:fallback /></xi:include>
+</book>
--- /dev/null
+<SECTION>
+<FILE>rtsp-address-pool</FILE>
+<TITLE>GstRTSPAddressPool</TITLE>
+
+<SUBSECTION Address>
+GST_RTSP_ADDRESS_POOL_ANY_IPV4
+GST_RTSP_ADDRESS_POOL_ANY_IPV6
+GstRTSPAddress
+GstRTSPAddressFlags
+gst_rtsp_address_copy
+gst_rtsp_address_free
+
+<SUBSECTION AddressPool>
+GstRTSPAddressPool
+GstRTSPAddressPoolClass
+GstRTSPAddressPoolResult
+gst_rtsp_address_pool_new
+gst_rtsp_address_pool_clear
+gst_rtsp_address_pool_dump
+gst_rtsp_address_pool_add_range
+gst_rtsp_address_pool_has_unicast_addresses
+gst_rtsp_address_pool_acquire_address
+gst_rtsp_address_pool_reserve_address
+<SUBSECTION Standard>
+GST_RTSP_ADDRESS_POOL_CAST
+GST_RTSP_ADDRESS_POOL_CLASS_CAST
+GST_IS_RTSP_ADDRESS_POOL
+GST_IS_RTSP_ADDRESS_POOL_CLASS
+GST_RTSP_ADDRESS_POOL
+GST_RTSP_ADDRESS_POOL_CLASS
+GST_RTSP_ADDRESS_POOL_GET_CLASS
+GST_TYPE_RTSP_ADDRESS_POOL
+GstRTSPAddressPoolPrivate
+gst_rtsp_address_get_type
+gst_rtsp_address_pool_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-auth</FILE>
+<TITLE>GstRTSPAuth</TITLE>
+GstRTSPAuth
+GstRTSPAuthClass
+gst_rtsp_auth_new
+
+gst_rtsp_auth_get_tls_certificate
+gst_rtsp_auth_set_tls_certificate
+gst_rtsp_auth_make_basic
+gst_rtsp_auth_add_basic
+gst_rtsp_auth_remove_basic
+gst_rtsp_auth_check
+gst_rtsp_auth_get_default_token
+gst_rtsp_auth_set_default_token
+
+<SUBSECTION AuthChecks>
+GST_RTSP_AUTH_CHECK_CONNECT
+GST_RTSP_AUTH_CHECK_URL
+GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS
+GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT
+GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS
+
+<SUBSECTION AuthTokens>
+GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE
+GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS
+
+<SUBSECTION AuthPermissions>
+GST_RTSP_PERM_MEDIA_FACTORY_ACCESS
+GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT
+<SUBSECTION Standard>
+GST_RTSP_AUTH_CAST
+GST_RTSP_AUTH_CLASS_CAST
+GST_IS_RTSP_AUTH
+GST_IS_RTSP_AUTH_CLASS
+GST_RTSP_AUTH
+GST_RTSP_AUTH_CLASS
+GST_RTSP_AUTH_GET_CLASS
+GST_TYPE_RTSP_AUTH
+GstRTSPAuthPrivate
+gst_rtsp_auth_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-client</FILE>
+<TITLE>GstRTSPClient</TITLE>
+<SUBSECTION Client>
+GstRTSPClient
+GstRTSPClientClass
+
+gst_rtsp_client_new
+gst_rtsp_client_close
+
+gst_rtsp_client_get_session_pool
+gst_rtsp_client_set_session_pool
+
+gst_rtsp_client_get_mount_points
+gst_rtsp_client_set_mount_points
+
+gst_rtsp_client_get_auth
+gst_rtsp_client_set_auth
+
+gst_rtsp_client_get_thread_pool
+gst_rtsp_client_set_thread_pool
+
+gst_rtsp_client_get_connection
+gst_rtsp_client_set_connection
+
+gst_rtsp_client_attach
+
+GstRTSPClientSendFunc
+gst_rtsp_client_set_send_func
+
+gst_rtsp_client_handle_message
+gst_rtsp_client_send_message
+
+GstRTSPClientSessionFilterFunc
+gst_rtsp_client_session_filter
+<SUBSECTION Standard>
+GST_RTSP_CLIENT_CAST
+GST_RTSP_CLIENT_CLASS_CAST
+GST_IS_RTSP_CLIENT
+GST_IS_RTSP_CLIENT_CLASS
+GST_RTSP_CLIENT
+GST_RTSP_CLIENT_CLASS
+GST_RTSP_CLIENT_GET_CLASS
+GST_TYPE_RTSP_CLIENT
+GstRTSPClientPrivate
+gst_rtsp_client_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-context</FILE>
+<TITLE>GstRTSPContext</TITLE>
+GstRTSPContext
+gst_rtsp_context_get_current
+gst_rtsp_context_push_current
+gst_rtsp_context_pop_current
+<SUBSECTION Standard>
+GST_TYPE_RTSP_CONTEXT
+gst_rtsp_context_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-media</FILE>
+<TITLE>GstRTSPMedia</TITLE>
+GstRTSPMedia
+GstRTSPMediaClass
+gst_rtsp_media_new
+gst_rtsp_media_get_element
+gst_rtsp_media_take_pipeline
+
+gst_rtsp_media_set_permissions
+gst_rtsp_media_get_permissions
+
+gst_rtsp_media_set_shared
+gst_rtsp_media_is_shared
+
+gst_rtsp_media_set_reusable
+gst_rtsp_media_is_reusable
+
+gst_rtsp_media_set_profiles
+gst_rtsp_media_get_profiles
+
+gst_rtsp_media_set_protocols
+gst_rtsp_media_get_protocols
+
+gst_rtsp_media_set_eos_shutdown
+gst_rtsp_media_is_eos_shutdown
+
+gst_rtsp_media_set_address_pool
+gst_rtsp_media_get_address_pool
+
+gst_rtsp_media_set_buffer_size
+gst_rtsp_media_get_buffer_size
+
+gst_rtsp_media_setup_sdp
+
+<SUBSECTION MediaPrepare>
+gst_rtsp_media_prepare
+gst_rtsp_media_unprepare
+GstRTSPMediaStatus
+gst_rtsp_media_get_status
+
+<SUBSECTION MediaSuspend>
+gst_rtsp_media_set_suspend_mode
+gst_rtsp_media_get_suspend_mode
+
+GstRTSPSuspendMode
+gst_rtsp_media_suspend
+gst_rtsp_media_unsuspend
+
+<SUBSECTION MediaStreams>
+gst_rtsp_media_collect_streams
+gst_rtsp_media_create_stream
+gst_rtsp_media_n_streams
+gst_rtsp_media_get_stream
+gst_rtsp_media_find_stream
+
+<SUBSECTION MediaState>
+gst_rtsp_media_seek
+gst_rtsp_media_get_range_string
+
+gst_rtsp_media_set_state
+gst_rtsp_media_set_pipeline_state
+
+<SUBSECTION MediaClocks>
+gst_rtsp_media_get_clock
+gst_rtsp_media_get_base_time
+gst_rtsp_media_use_time_provider
+gst_rtsp_media_is_time_provider
+gst_rtsp_media_get_time_provider
+<SUBSECTION Standard>
+GST_RTSP_MEDIA_CAST
+GST_RTSP_MEDIA_CLASS_CAST
+GST_IS_RTSP_MEDIA
+GST_IS_RTSP_MEDIA_CLASS
+GST_RTSP_MEDIA
+GST_RTSP_MEDIA_CLASS
+GST_RTSP_MEDIA_GET_CLASS
+GST_TYPE_RTSP_MEDIA
+GstRTSPMediaPrivate
+gst_rtsp_media_get_type
+GST_TYPE_RTSP_SUSPEND_MODE
+gst_rtsp_suspend_mode_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-media-factory</FILE>
+<TITLE>GstRTSPMediaFactory</TITLE>
+GstRTSPMediaFactory
+GstRTSPMediaFactoryClass
+gst_rtsp_media_factory_new
+
+gst_rtsp_media_factory_get_launch
+gst_rtsp_media_factory_set_launch
+
+gst_rtsp_media_factory_get_permissions
+gst_rtsp_media_factory_set_permissions
+gst_rtsp_media_factory_add_role
+
+gst_rtsp_media_factory_set_shared
+gst_rtsp_media_factory_is_shared
+
+gst_rtsp_media_factory_is_eos_shutdown
+gst_rtsp_media_factory_set_eos_shutdown
+
+gst_rtsp_media_factory_get_protocols
+gst_rtsp_media_factory_set_protocols
+
+gst_rtsp_media_factory_set_profiles
+gst_rtsp_media_factory_get_profiles
+
+gst_rtsp_media_factory_get_address_pool
+gst_rtsp_media_factory_set_address_pool
+
+gst_rtsp_media_factory_get_buffer_size
+gst_rtsp_media_factory_set_buffer_size
+
+gst_rtsp_media_factory_get_suspend_mode
+gst_rtsp_media_factory_set_suspend_mode
+
+gst_rtsp_media_factory_construct
+gst_rtsp_media_factory_create_element
+
+<SUBSECTION Standard>
+GST_RTSP_MEDIA_FACTORY_CAST
+GST_RTSP_MEDIA_FACTORY_CLASS_CAST
+GST_IS_RTSP_MEDIA_FACTORY
+GST_IS_RTSP_MEDIA_FACTORY_CLASS
+GST_RTSP_MEDIA_FACTORY
+GST_RTSP_MEDIA_FACTORY_CLASS
+GST_RTSP_MEDIA_FACTORY_GET_CLASS
+GST_TYPE_RTSP_MEDIA_FACTORY
+GstRTSPMediaFactoryPrivate
+gst_rtsp_media_factory_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-media-factory-uri</FILE>
+<TITLE>GstRTSPMediaFactoryURI</TITLE>
+GstRTSPMediaFactoryURI
+GstRTSPMediaFactoryURIClass
+gst_rtsp_media_factory_uri_new
+gst_rtsp_media_factory_uri_set_uri
+gst_rtsp_media_factory_uri_get_uri
+<SUBSECTION Standard>
+GST_RTSP_MEDIA_FACTORY_URI_CAST
+GST_RTSP_MEDIA_FACTORY_URI_CLASS_CAST
+GST_IS_RTSP_MEDIA_FACTORY_URI
+GST_IS_RTSP_MEDIA_FACTORY_URI_CLASS
+GST_RTSP_MEDIA_FACTORY_URI
+GST_RTSP_MEDIA_FACTORY_URI_CLASS
+GST_RTSP_MEDIA_FACTORY_URI_GET_CLASS
+GST_TYPE_RTSP_MEDIA_FACTORY_URI
+GstRTSPMediaFactoryURIPrivate
+gst_rtsp_media_factory_uri_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-mount-points</FILE>
+<TITLE>GstRTSPMountPoints</TITLE>
+GstRTSPMountPoints
+GstRTSPMountPointsClass
+gst_rtsp_mount_points_new
+gst_rtsp_mount_points_add_factory
+gst_rtsp_mount_points_remove_factory
+gst_rtsp_mount_points_match
+gst_rtsp_mount_points_make_path
+<SUBSECTION Standard>
+GST_RTSP_MOUNT_POINTS_CAST
+GST_RTSP_MOUNT_POINTS_CLASS_CAST
+GST_IS_RTSP_MOUNT_POINTS
+GST_IS_RTSP_MOUNT_POINTS_CLASS
+GST_RTSP_MOUNT_POINTS
+GST_RTSP_MOUNT_POINTS_CLASS
+GST_RTSP_MOUNT_POINTS_GET_CLASS
+GST_TYPE_RTSP_MOUNT_POINTS
+GstRTSPMountPointsPrivate
+gst_rtsp_mount_points_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-params</FILE>
+<TITLE>GstRTSPParams</TITLE>
+gst_rtsp_params_get
+gst_rtsp_params_set
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-permissions</FILE>
+<TITLE>GstRTSPPermissions</TITLE>
+GstRTSPPermissions
+gst_rtsp_permissions_new
+gst_rtsp_permissions_ref
+gst_rtsp_permissions_unref
+gst_rtsp_permissions_add_role
+gst_rtsp_permissions_add_role_valist
+gst_rtsp_permissions_remove_role
+gst_rtsp_permissions_get_role
+gst_rtsp_permissions_is_allowed
+<SUBSECTION Standard>
+GST_RTSP_PERMISSIONS_CAST
+GST_IS_RTSP_PERMISSIONS
+GST_RTSP_PERMISSIONS
+GST_TYPE_RTSP_PERMISSIONS
+gst_rtsp_permissions_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-sdp</FILE>
+<TITLE>GstRTSPSdp</TITLE>
+GstSDPInfo
+gst_rtsp_sdp_from_media
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-server</FILE>
+<TITLE>GstRTSPServer</TITLE>
+GstRTSPServer
+GstRTSPServerClass
+
+gst_rtsp_server_new
+
+gst_rtsp_server_get_address
+gst_rtsp_server_set_address
+
+gst_rtsp_server_get_service
+gst_rtsp_server_set_service
+
+gst_rtsp_server_get_backlog
+gst_rtsp_server_set_backlog
+
+gst_rtsp_server_get_bound_port
+
+gst_rtsp_server_get_mount_points
+gst_rtsp_server_set_mount_points
+
+gst_rtsp_server_get_session_pool
+gst_rtsp_server_set_session_pool
+
+gst_rtsp_server_get_thread_pool
+gst_rtsp_server_set_thread_pool
+
+gst_rtsp_server_get_auth
+gst_rtsp_server_set_auth
+
+gst_rtsp_server_transfer_connection
+gst_rtsp_server_io_func
+gst_rtsp_server_create_socket
+gst_rtsp_server_create_source
+gst_rtsp_server_attach
+
+GstRTSPServerClientFilterFunc
+gst_rtsp_server_client_filter
+
+<SUBSECTION Standard>
+GST_IS_RTSP_SERVER
+GST_RTSP_SERVER_CAST
+GST_RTSP_SERVER_CLASS_CAST
+GST_IS_RTSP_SERVER_CLASS
+GST_RTSP_SERVER
+GST_RTSP_SERVER_CLASS
+GST_RTSP_SERVER_GET_CLASS
+GST_TYPE_RTSP_SERVER
+GstRTSPServerPrivate
+gst_rtsp_server_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-session</FILE>
+<TITLE>GstRTSPSession</TITLE>
+GstRTSPSession
+GstRTSPSessionClass
+
+gst_rtsp_session_new
+gst_rtsp_session_get_sessionid
+
+gst_rtsp_session_get_header
+
+gst_rtsp_session_set_timeout
+gst_rtsp_session_get_timeout
+
+gst_rtsp_session_touch
+gst_rtsp_session_prevent_expire
+gst_rtsp_session_allow_expire
+gst_rtsp_session_next_timeout
+gst_rtsp_session_is_expired
+
+gst_rtsp_session_manage_media
+gst_rtsp_session_release_media
+
+gst_rtsp_session_get_media
+
+GstRTSPFilterResult
+GstRTSPSessionFilterFunc
+gst_rtsp_session_filter
+<SUBSECTION Standard>
+GST_RTSP_SESSION_CAST
+GST_RTSP_SESSION_CLASS_CAST
+GST_IS_RTSP_SESSION
+GST_IS_RTSP_SESSION_CLASS
+GST_RTSP_SESSION
+GST_RTSP_SESSION_CLASS
+GST_RTSP_SESSION_GET_CLASS
+GST_TYPE_RTSP_SESSION
+GstRTSPSessionPrivate
+gst_rtsp_session_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-session-media</FILE>
+<TITLE>GstRTSPSessionMedia</TITLE>
+GstRTSPSessionMedia
+GstRTSPSessionMediaClass
+gst_rtsp_session_media_new
+gst_rtsp_session_media_matches
+
+gst_rtsp_session_media_get_media
+gst_rtsp_session_media_get_base_time
+gst_rtsp_session_media_get_rtpinfo
+
+gst_rtsp_session_media_set_state
+
+gst_rtsp_session_media_get_rtsp_state
+gst_rtsp_session_media_set_rtsp_state
+
+gst_rtsp_session_media_get_transport
+gst_rtsp_session_media_set_transport
+
+gst_rtsp_session_media_alloc_channels
+<SUBSECTION Standard>
+GST_RTSP_SESSION_MEDIA_CAST
+GST_RTSP_SESSION_MEDIA_CLASS_CAST
+GST_IS_RTSP_SESSION_MEDIA
+GST_IS_RTSP_SESSION_MEDIA_CLASS
+GST_RTSP_SESSION_MEDIA
+GST_RTSP_SESSION_MEDIA_CLASS
+GST_RTSP_SESSION_MEDIA_GET_CLASS
+GST_TYPE_RTSP_SESSION_MEDIA
+GstRTSPSessionMediaPrivate
+gst_rtsp_session_media_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-session-pool</FILE>
+<TITLE>GstRTSPSessionPool</TITLE>
+GstRTSPSessionPool
+GstRTSPSessionPoolClass
+gst_rtsp_session_pool_new
+
+gst_rtsp_session_pool_get_max_sessions
+gst_rtsp_session_pool_set_max_sessions
+
+gst_rtsp_session_pool_get_n_sessions
+
+gst_rtsp_session_pool_create
+gst_rtsp_session_pool_find
+gst_rtsp_session_pool_remove
+
+gst_rtsp_session_pool_cleanup
+
+GstRTSPSessionPoolFunc
+gst_rtsp_session_pool_create_watch
+
+GstRTSPSessionPoolFilterFunc
+gst_rtsp_session_pool_filter
+<SUBSECTION Standard>
+GST_RTSP_SESSION_POOL_CAST
+GST_RTSP_SESSION_POOL_CLASS_CAST
+GST_IS_RTSP_SESSION_POOL
+GST_IS_RTSP_SESSION_POOL_CLASS
+GST_RTSP_SESSION_POOL
+GST_RTSP_SESSION_POOL_CLASS
+GST_RTSP_SESSION_POOL_GET_CLASS
+GST_TYPE_RTSP_SESSION_POOL
+GstRTSPSessionPoolPrivate
+gst_rtsp_session_pool_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-stream</FILE>
+<TITLE>GstRTSPStream</TITLE>
+GstRTSPStream
+GstRTSPStreamClass
+gst_rtsp_stream_new
+
+gst_rtsp_stream_get_index
+gst_rtsp_stream_get_srcpad
+
+gst_rtsp_stream_get_control
+gst_rtsp_stream_set_control
+gst_rtsp_stream_has_control
+
+gst_rtsp_stream_get_mtu
+gst_rtsp_stream_set_mtu
+
+gst_rtsp_stream_get_dscp_qos
+gst_rtsp_stream_set_dscp_qos
+
+gst_rtsp_stream_set_profiles
+gst_rtsp_stream_get_profiles
+
+gst_rtsp_stream_get_protocols
+gst_rtsp_stream_set_protocols
+
+gst_rtsp_stream_is_transport_supported
+
+gst_rtsp_stream_get_address_pool
+gst_rtsp_stream_set_address_pool
+gst_rtsp_stream_reserve_address
+
+gst_rtsp_stream_join_bin
+gst_rtsp_stream_leave_bin
+
+gst_rtsp_stream_get_server_port
+gst_rtsp_stream_get_multicast_address
+gst_rtsp_stream_get_rtpsession
+gst_rtsp_stream_get_ssrc
+gst_rtsp_stream_get_rtpinfo
+gst_rtsp_stream_get_caps
+gst_rtsp_stream_get_pt
+
+gst_rtsp_stream_recv_rtcp
+gst_rtsp_stream_recv_rtp
+
+gst_rtsp_stream_add_transport
+gst_rtsp_stream_remove_transport
+
+gst_rtsp_stream_get_rtp_socket
+gst_rtsp_stream_get_rtcp_socket
+
+gst_rtsp_stream_set_blocked
+gst_rtsp_stream_is_blocking
+
+gst_rtsp_stream_update_crypto
+
+GstRTSPStreamTransportFilterFunc
+gst_rtsp_stream_transport_filter
+
+<SUBSECTION Standard>
+GST_RTSP_STREAM_CAST
+GST_RTSP_STREAM_CLASS_CAST
+GST_IS_RTSP_STREAM
+GST_IS_RTSP_STREAM_CLASS
+GST_RTSP_STREAM
+GST_RTSP_STREAM_CLASS
+GST_RTSP_STREAM_GET_CLASS
+GST_TYPE_RTSP_STREAM
+GstRTSPStreamPrivate
+gst_rtsp_stream_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-stream-transport</FILE>
+<TITLE>GstRTSPStreamTransport</TITLE>
+GstRTSPStreamTransport
+GstRTSPStreamTransportClass
+gst_rtsp_stream_transport_new
+
+gst_rtsp_stream_transport_get_stream
+
+gst_rtsp_stream_transport_get_transport
+gst_rtsp_stream_transport_set_transport
+
+gst_rtsp_stream_transport_get_url
+gst_rtsp_stream_transport_set_url
+
+gst_rtsp_stream_transport_get_rtpinfo
+
+GstRTSPSendFunc
+gst_rtsp_stream_transport_set_callbacks
+
+GstRTSPKeepAliveFunc
+gst_rtsp_stream_transport_set_keepalive
+gst_rtsp_stream_transport_keep_alive
+
+gst_rtsp_stream_transport_set_active
+
+gst_rtsp_stream_transport_set_timed_out
+gst_rtsp_stream_transport_is_timed_out
+
+gst_rtsp_stream_transport_send_rtcp
+gst_rtsp_stream_transport_send_rtp
+
+<SUBSECTION Standard>
+GST_RTSP_STREAM_TRANSPORT_CAST
+GST_RTSP_STREAM_TRANSPORT_CLASS_CAST
+GST_IS_RTSP_STREAM_TRANSPORT
+GST_IS_RTSP_STREAM_TRANSPORT_CLASS
+GST_RTSP_STREAM_TRANSPORT
+GST_RTSP_STREAM_TRANSPORT_CLASS
+GST_RTSP_STREAM_TRANSPORT_GET_CLASS
+GST_TYPE_RTSP_STREAM_TRANSPORT
+GstRTSPStreamTransportPrivate
+gst_rtsp_stream_transport_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-thread-pool</FILE>
+<TITLE>GstRTSPThreadPool</TITLE>
+<SUBSECTION Thread>
+GstRTSPThreadType
+GstRTSPThread
+
+gst_rtsp_thread_new
+gst_rtsp_thread_ref
+gst_rtsp_thread_unref
+gst_rtsp_thread_reuse
+gst_rtsp_thread_stop
+
+<SUBSECTION ThreadPool>
+GstRTSPThreadPool
+GstRTSPThreadPoolClass
+gst_rtsp_thread_pool_new
+
+gst_rtsp_thread_pool_get_max_threads
+gst_rtsp_thread_pool_set_max_threads
+
+gst_rtsp_thread_pool_get_thread
+gst_rtsp_thread_pool_cleanup
+<SUBSECTION Standard>
+GST_RTSP_THREAD_CAST
+GST_RTSP_THREAD_POOL_CAST
+GST_RTSP_THREAD_POOL_CLASS_CAST
+GST_IS_RTSP_THREAD
+GST_IS_RTSP_THREAD_POOL
+GST_IS_RTSP_THREAD_POOL_CLASS
+GST_RTSP_THREAD
+GST_RTSP_THREAD_POOL
+GST_RTSP_THREAD_POOL_CLASS
+GST_RTSP_THREAD_POOL_GET_CLASS
+GST_TYPE_RTSP_THREAD
+GST_TYPE_RTSP_THREAD_POOL
+GstRTSPThreadPoolPrivate
+gst_rtsp_thread_get_type
+gst_rtsp_thread_pool_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-token</FILE>
+<TITLE>GstRTSPToken</TITLE>
+GstRTSPToken
+gst_rtsp_token_new_empty
+gst_rtsp_token_new
+gst_rtsp_token_new_valist
+gst_rtsp_token_ref
+gst_rtsp_token_unref
+gst_rtsp_token_get_structure
+gst_rtsp_token_writable_structure
+gst_rtsp_token_get_string
+gst_rtsp_token_is_allowed
+<SUBSECTION Standard>
+GST_RTSP_TOKEN_CAST
+GST_IS_RTSP_TOKEN
+GST_RTSP_TOKEN
+GST_TYPE_RTSP_TOKEN
+gst_rtsp_token_get_type
+</SECTION>
+
--- /dev/null
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-auth.h>
+gst_rtsp_auth_get_type
+
+#include <gst/rtsp-server/rtsp-mount-points.h>
+gst_rtsp_mount_points_get_type
+
+#include <gst/rtsp-server/rtsp-media-factory.h>
+gst_rtsp_media_factory_get_type
+
+#include <gst/rtsp-server/rtsp-media.h>
+gst_rtsp_media_get_type
+
+#include <gst/rtsp-server/rtsp-server.h>
+gst_rtsp_server_get_type
+
+#include <gst/rtsp-server/rtsp-session-pool.h>
+gst_rtsp_session_pool_get_type
+
+#include <gst/rtsp-server/rtsp-session.h>
+gst_rtsp_session_get_type
+
+#include <gst/rtsp-server/rtsp-client.h>
+gst_rtsp_client_get_type
--- /dev/null
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
+<html>
+<head>
+<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
+<title>GStreamer RTSP Server Reference Manual: GstRTSPAuth</title>
+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
+<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
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+<td width="100%" align="left" class="shortcuts">
+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#GstRTSPAuth.description" class="shortcut">Description</a></span><span id="nav_hierarchy"> <span class="dim">|</span>
+ <a href="#GstRTSPAuth.object-hierarchy" class="shortcut">Object Hierarchy</a></span>
+</td>
+<td><a accesskey="h" href="index.html"><img src="home.png" width="16" height="16" border="0" alt="Home"></a></td>
+<td><a accesskey="u" href="ch01.html"><img src="up.png" width="16" height="16" border="0" alt="Up"></a></td>
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+</tr></table>
+<div class="refentry">
+<a name="GstRTSPAuth"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="GstRTSPAuth.top_of_page"></a>GstRTSPAuth</span></h2>
+<p>GstRTSPAuth — Authentication and authorization</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="GstRTSPAuth.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-new" title="gst_rtsp_auth_new ()">gst_rtsp_auth_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="returnvalue">GTlsCertificate</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-get-tls-certificate" title="gst_rtsp_auth_get_tls_certificate ()">gst_rtsp_auth_get_tls_certificate</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate" title="gst_rtsp_auth_set_tls_certificate ()">gst_rtsp_auth_set_tls_certificate</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-make-basic" title="gst_rtsp_auth_make_basic ()">gst_rtsp_auth_make_basic</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-add-basic" title="gst_rtsp_auth_add_basic ()">gst_rtsp_auth_add_basic</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-remove-basic" title="gst_rtsp_auth_remove_basic ()">gst_rtsp_auth_remove_basic</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-check" title="gst_rtsp_auth_check ()">gst_rtsp_auth_check</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-get-default-token" title="gst_rtsp_auth_get_default_token ()">gst_rtsp_auth_get_default_token</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-default-token" title="gst_rtsp_auth_set_default_token ()">gst_rtsp_auth_set_default_token</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPAuth.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GstRTSPAuth-struct" title="struct GstRTSPAuth">GstRTSPAuth</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GstRTSPAuthClass" title="struct GstRTSPAuthClass">GstRTSPAuthClass</a></td>
+</tr>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-CONNECT:CAPS" title="GST_RTSP_AUTH_CHECK_CONNECT">GST_RTSP_AUTH_CHECK_CONNECT</a></td>
+</tr>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-URL:CAPS" title="GST_RTSP_AUTH_CHECK_URL">GST_RTSP_AUTH_CHECK_URL</a></td>
+</tr>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-ACCESS:CAPS" title="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS">GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS</a></td>
+</tr>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-CONSTRUCT:CAPS" title="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT">GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT</a></td>
+</tr>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-TRANSPORT-CLIENT-SETTINGS:CAPS" title="GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS">GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS</a></td>
+</tr>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-TOKEN-MEDIA-FACTORY-ROLE:CAPS" title="GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE">GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE</a></td>
+</tr>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-TOKEN-TRANSPORT-CLIENT-SETTINGS:CAPS" title="GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS">GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS</a></td>
+</tr>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-ACCESS:CAPS" title="GST_RTSP_PERM_MEDIA_FACTORY_ACCESS">GST_RTSP_PERM_MEDIA_FACTORY_ACCESS</a></td>
+</tr>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-CONSTRUCT:CAPS" title="GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT">GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPAuth.object-hierarchy"></a><h2>Object Hierarchy</h2>
+<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+ <span class="lineart">╰──</span> GstRTSPAuth
+</pre>
+</div>
+<div class="refsect1">
+<a name="GstRTSPAuth.description"></a><h2>Description</h2>
+<p>The <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> object is responsible for checking if the current user is
+allowed to perform requested actions. The default implementation has some
+reasonable checks but subclasses can implement custom security policies.</p>
+<p>A new auth object is made with <a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-new" title="gst_rtsp_auth_new ()"><code class="function">gst_rtsp_auth_new()</code></a>. It is usually configured
+on the <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> object.</p>
+<p>The RTSP server will call <a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-check" title="gst_rtsp_auth_check ()"><code class="function">gst_rtsp_auth_check()</code></a> with a string describing the
+check to perform. The possible checks are prefixed with
+GST_RTSP_AUTH_CHECK_*. Depending on the check, the default implementation
+will use the current <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> and
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> on the object to check if an operation is allowed.</p>
+<p>The default <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> object has support for basic authentication. With
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-add-basic" title="gst_rtsp_auth_add_basic ()"><code class="function">gst_rtsp_auth_add_basic()</code></a> you can add a basic authentication string together
+with the <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> that will become active when successfully
+authenticated.</p>
+<p>When a TLS certificate has been set with <a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate" title="gst_rtsp_auth_set_tls_certificate ()"><code class="function">gst_rtsp_auth_set_tls_certificate()</code></a>,
+the default auth object will require the client to connect with a TLS
+connection.</p>
+<p>Last reviewed on 2013-07-16 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="GstRTSPAuth.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-auth-new"></a><h3>gst_rtsp_auth_new ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
+gst_rtsp_auth_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Create a new <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> instance.</p>
+<div class="refsect3">
+<a name="id-1.2.16.7.2.5"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-auth-get-tls-certificate"></a><h3>gst_rtsp_auth_get_tls_certificate ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="returnvalue">GTlsCertificate</span></a> *
+gst_rtsp_auth_get_tls_certificate (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>);</pre>
+<p>Get the <a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="type">GTlsCertificate</span></a> used for negotiating TLS <em class="parameter"><code>auth</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.16.7.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>auth</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.16.7.3.6"></a><h4>Returns</h4>
+<p> the <a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="type">GTlsCertificate</span></a> of <em class="parameter"><code>auth</code></em>
+. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-auth-set-tls-certificate"></a><h3>gst_rtsp_auth_set_tls_certificate ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_auth_set_tls_certificate (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="type">GTlsCertificate</span></a> *cert</code></em>);</pre>
+<p>Set the TLS certificate for the auth. Client connections will only
+be accepted when TLS is negotiated.</p>
+<div class="refsect3">
+<a name="id-1.2.16.7.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>auth</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>cert</p></td>
+<td class="parameter_description"><p> a <a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="type">GTlsCertificate</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-auth-make-basic"></a><h3>gst_rtsp_auth_make_basic ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_auth_make_basic (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *user</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *pass</code></em>);</pre>
+<p>Construct a Basic authorisation token from <em class="parameter"><code>user</code></em>
+ and <em class="parameter"><code>pass</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.16.7.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>user</p></td>
+<td class="parameter_description"><p>a userid</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>pass</p></td>
+<td class="parameter_description"><p>a password</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.16.7.5.6"></a><h4>Returns</h4>
+<p> the base64 encoding of the string <em class="parameter"><code>user</code></em>
+:<em class="parameter"><code>pass</code></em>
+.
+<a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-auth-add-basic"></a><h3>gst_rtsp_auth_add_basic ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_auth_add_basic (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *basic</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
+<p>Add a basic token for the default authentication algorithm that
+enables the client with privileges listed in <em class="parameter"><code>token</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.16.7.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>auth</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>basic</p></td>
+<td class="parameter_description"><p>the basic token</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>token</p></td>
+<td class="parameter_description"><p> authorisation token. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-auth-remove-basic"></a><h3>gst_rtsp_auth_remove_basic ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_auth_remove_basic (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *basic</code></em>);</pre>
+<p>Add a basic token for the default authentication algorithm that
+enables the client with privileges from <em class="parameter"><code>authgroup</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.16.7.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>auth</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>basic</p></td>
+<td class="parameter_description"><p> the basic token. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-auth-check"></a><h3>gst_rtsp_auth_check ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_auth_check (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *check</code></em>);</pre>
+<p>Check if <em class="parameter"><code>check</code></em>
+ is allowed in the current context.</p>
+<div class="refsect3">
+<a name="id-1.2.16.7.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>check</p></td>
+<td class="parameter_description"><p>the item to check</p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.16.7.8.6"></a><h4>Returns</h4>
+<p> FALSE if check failed.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-auth-get-default-token"></a><h3>gst_rtsp_auth_get_default_token ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
+gst_rtsp_auth_get_default_token (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>);</pre>
+<p>Get the default token for <em class="parameter"><code>auth</code></em>
+. This token will be used for unauthenticated
+users.</p>
+<div class="refsect3">
+<a name="id-1.2.16.7.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>auth</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.16.7.9.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> of <em class="parameter"><code>auth</code></em>
+. <a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-unref" title="gst_rtsp_token_unref ()"><code class="function">gst_rtsp_token_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-auth-set-default-token"></a><h3>gst_rtsp_auth_set_default_token ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_auth_set_default_token (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
+<p>Set the default <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> to <em class="parameter"><code>token</code></em>
+ in <em class="parameter"><code>auth</code></em>
+. The default token will
+be used for unauthenticated users.</p>
+<div class="refsect3">
+<a name="id-1.2.16.7.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>auth</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>token</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPAuth.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPAuth-struct"></a><h3>struct GstRTSPAuth</h3>
+<pre class="programlisting">struct GstRTSPAuth;</pre>
+<p>The authentication structure.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPAuthClass"></a><h3>struct GstRTSPAuthClass</h3>
+<pre class="programlisting">struct GstRTSPAuthClass {
+ GObjectClass parent_class;
+
+ gboolean (*authenticate) (GstRTSPAuth *auth, GstRTSPContext *ctx);
+ gboolean (*check) (GstRTSPAuth *auth, GstRTSPContext *ctx,
+ const gchar *check);
+};
+</pre>
+<p>The authentication class.</p>
+<div class="refsect3">
+<a name="id-1.2.16.8.3.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPAuthClass.parent-class"></a>parent_class</code></em>;</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPAuthClass.authenticate"></a>authenticate</code></em> ()</p></td>
+<td class="struct_member_description"><p>check the authentication of a client. The default implementation
+checks if the authentication in the header matches one of the basic
+authentication tokens. This function should set the authgroup field
+in the context.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPAuthClass.check"></a>check</code></em> ()</p></td>
+<td class="struct_member_description"><p>check if a resource can be accessed. this function should
+call authenticate to authenticate the client when needed. The method
+should also construct and send an appropriate response message on
+error.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GST-RTSP-AUTH-CHECK-CONNECT:CAPS"></a><h3>GST_RTSP_AUTH_CHECK_CONNECT</h3>
+<pre class="programlisting">#define GST_RTSP_AUTH_CHECK_CONNECT "auth.check.connect"
+</pre>
+<p>Check a new connection</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GST-RTSP-AUTH-CHECK-URL:CAPS"></a><h3>GST_RTSP_AUTH_CHECK_URL</h3>
+<pre class="programlisting">#define GST_RTSP_AUTH_CHECK_URL "auth.check.url"
+</pre>
+<p>Check the URL and methods</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-ACCESS:CAPS"></a><h3>GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS</h3>
+<pre class="programlisting">#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS "auth.check.media.factory.access"
+</pre>
+<p>Check if access is allowed to a factory.
+When access is not allowed an 404 Not Found is sent in the response.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-CONSTRUCT:CAPS"></a><h3>GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT</h3>
+<pre class="programlisting">#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT "auth.check.media.factory.construct"
+</pre>
+<p>Check if media can be constructed from a media factory
+A response should be sent on error.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GST-RTSP-AUTH-CHECK-TRANSPORT-CLIENT-SETTINGS:CAPS"></a><h3>GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS</h3>
+<pre class="programlisting">#define GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS "auth.check.transport.client-settings"
+</pre>
+<p>Check if the client can specify TTL, destination and
+port pair in multicast. No response is sent when the check returns
+<a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#FALSE:CAPS"><code class="literal">FALSE</code></a>.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GST-RTSP-TOKEN-MEDIA-FACTORY-ROLE:CAPS"></a><h3>GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE</h3>
+<pre class="programlisting">#define GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE "media.factory.role"
+</pre>
+<p>G_TYPE_STRING, the role to use when dealing with media factories</p>
+<p>The default <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> object uses this string in the token to find the
+role of the media factory. It will then retrieve the <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> of
+the media factory and retrieve the role with the same name.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GST-RTSP-TOKEN-TRANSPORT-CLIENT-SETTINGS:CAPS"></a><h3>GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS</h3>
+<pre class="programlisting">#define GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS "transport.client-settings"
+</pre>
+<p>G_TYPE_BOOLEAN, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the client can specify TTL, destination and
+ port pair in multicast.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GST-RTSP-PERM-MEDIA-FACTORY-ACCESS:CAPS"></a><h3>GST_RTSP_PERM_MEDIA_FACTORY_ACCESS</h3>
+<pre class="programlisting">#define GST_RTSP_PERM_MEDIA_FACTORY_ACCESS "media.factory.access"
+</pre>
+<p>G_TYPE_BOOLEAN, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media can be accessed, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#FALSE:CAPS"><code class="literal">FALSE</code></a> will
+return a 404 Not Found error when trying to access the media.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GST-RTSP-PERM-MEDIA-FACTORY-CONSTRUCT:CAPS"></a><h3>GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT</h3>
+<pre class="programlisting">#define GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT "media.factory.construct"
+</pre>
+<p>G_TYPE_BOOLEAN, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media can be constructed, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#FALSE:CAPS"><code class="literal">FALSE</code></a> will
+return a 404 Not Found error when trying to access the media.</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPAuth.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
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+</html>
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+<td width="100%" align="left" class="shortcuts">
+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#GstRTSPClient.description" class="shortcut">Description</a></span><span id="nav_hierarchy"> <span class="dim">|</span>
+ <a href="#GstRTSPClient.object-hierarchy" class="shortcut">Object Hierarchy</a></span><span id="nav_properties"> <span class="dim">|</span>
+ <a href="#GstRTSPClient.properties" class="shortcut">Properties</a></span><span id="nav_signals"> <span class="dim">|</span>
+ <a href="#GstRTSPClient.signals" class="shortcut">Signals</a></span>
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+</tr></table>
+<div class="refentry">
+<a name="GstRTSPClient"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="GstRTSPClient.top_of_page"></a>GstRTSPClient</span></h2>
+<p>GstRTSPClient — A client connection state</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="GstRTSPClient.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="returnvalue">GstRTSPClient</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-new" title="gst_rtsp_client_new ()">gst_rtsp_client_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-close" title="gst_rtsp_client_close ()">gst_rtsp_client_close</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-session-pool" title="gst_rtsp_client_get_session_pool ()">gst_rtsp_client_get_session_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-session-pool" title="gst_rtsp_client_set_session_pool ()">gst_rtsp_client_set_session_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-mount-points" title="gst_rtsp_client_get_mount_points ()">gst_rtsp_client_get_mount_points</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-mount-points" title="gst_rtsp_client_set_mount_points ()">gst_rtsp_client_set_mount_points</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-auth" title="gst_rtsp_client_get_auth ()">gst_rtsp_client_get_auth</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-auth" title="gst_rtsp_client_set_auth ()">gst_rtsp_client_set_auth</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-thread-pool" title="gst_rtsp_client_get_thread_pool ()">gst_rtsp_client_get_thread_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-thread-pool" title="gst_rtsp_client_set_thread_pool ()">gst_rtsp_client_set_thread_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="returnvalue">GstRTSPConnection</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-connection" title="gst_rtsp_client_get_connection ()">gst_rtsp_client_get_connection</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-connection" title="gst_rtsp_client_set_connection ()">gst_rtsp_client_set_connection</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-attach" title="gst_rtsp_client_attach ()">gst_rtsp_client_attach</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<span class="c_punctuation">(</span><a class="link" href="GstRTSPClient.html#GstRTSPClientSendFunc" title="GstRTSPClientSendFunc ()">*GstRTSPClientSendFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-send-func" title="gst_rtsp_client_set_send_func ()">gst_rtsp_client_set_send_func</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-handle-message" title="gst_rtsp_client_handle_message ()">gst_rtsp_client_handle_message</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-send-message" title="gst_rtsp_client_send_message ()">gst_rtsp_client_send_message</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
+</td>
+<td class="function_name">
+<span class="c_punctuation">(</span><a class="link" href="GstRTSPClient.html#GstRTSPClientSessionFilterFunc" title="GstRTSPClientSessionFilterFunc ()">*GstRTSPClientSessionFilterFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()">gst_rtsp_client_session_filter</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPClient.properties"></a><h2>Properties</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="properties_type">
+<col width="300px" class="properties_name">
+<col width="200px" class="properties_flags">
+</colgroup>
+<tbody>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient--drop-backlog" title="The “drop-backlog” property">drop-backlog</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type">
+<a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *</td>
+<td class="property_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient--mount-points" title="The “mount-points” property">mount-points</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type">
+<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *</td>
+<td class="property_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient--session-pool" title="The “session-pool” property">session-pool</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPClient.signals"></a><h2>Signals</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="signals_return">
+<col width="300px" class="signals_name">
+<col width="200px" class="signals_flags">
+</colgroup>
+<tbody>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-closed" title="The “closed” signal">closed</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-describe-request" title="The “describe-request” signal">describe-request</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-get-parameter-request" title="The “get-parameter-request” signal">get-parameter-request</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-handle-response" title="The “handle-response” signal">handle-response</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-new-session" title="The “new-session” signal">new-session</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-options-request" title="The “options-request” signal">options-request</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-pause-request" title="The “pause-request” signal">pause-request</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-play-request" title="The “play-request” signal">play-request</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-send-message" title="The “send-message” signal">send-message</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-set-parameter-request" title="The “set-parameter-request” signal">set-parameter-request</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-setup-request" title="The “setup-request” signal">setup-request</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-teardown-request" title="The “teardown-request” signal">teardown-request</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPClient.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-struct" title="struct GstRTSPClient">GstRTSPClient</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPClient.html#GstRTSPClientClass" title="struct GstRTSPClientClass">GstRTSPClientClass</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPClient.object-hierarchy"></a><h2>Object Hierarchy</h2>
+<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+ <span class="lineart">╰──</span> GstRTSPClient
+</pre>
+</div>
+<div class="refsect1">
+<a name="GstRTSPClient.description"></a><h2>Description</h2>
+<p>The client object handles the connection with a client for as long as a TCP
+connection is open.</p>
+<p>A <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> is created by <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> when a new connection is
+accepted and it inherits the <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>, <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>,
+<a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> and <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> from the server.</p>
+<p>The client connection should be configured with the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> using
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-connection" title="gst_rtsp_client_set_connection ()"><code class="function">gst_rtsp_client_set_connection()</code></a> before it can be attached to a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a>
+using <a class="link" href="GstRTSPClient.html#gst-rtsp-client-attach" title="gst_rtsp_client_attach ()"><code class="function">gst_rtsp_client_attach()</code></a>. From then on the client will handle requests
+on the connection.</p>
+<p>Use <a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()"><code class="function">gst_rtsp_client_session_filter()</code></a> to iterate or modify all the
+<a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> objects managed by the client object.</p>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="GstRTSPClient.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-client-new"></a><h3>gst_rtsp_client_new ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="returnvalue">GstRTSPClient</span></a> *
+gst_rtsp_client_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Create a new <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> instance.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.2.5"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-close"></a><h3>gst_rtsp_client_close ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_client_close (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
+<p>Close the connection of <em class="parameter"><code>client</code></em>
+ and remove all media it was managing.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<p class="since">Since 1.4</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-get-session-pool"></a><h3>gst_rtsp_client_get_session_pool ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
+gst_rtsp_client_get_session_pool (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
+<p>Get the <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> object that <em class="parameter"><code>client</code></em>
+ uses to manage its sessions.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.4.6"></a><h4>Returns</h4>
+<p> a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>, unref after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-set-session-pool"></a><h3>gst_rtsp_client_set_session_pool ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_client_set_session_pool (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
+<p>Set <em class="parameter"><code>pool</code></em>
+ as the sessionpool for <em class="parameter"><code>client</code></em>
+ which it will use to find
+or allocate sessions. the sessionpool is usually inherited from the server
+that created the client but can be overridden later.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-get-mount-points"></a><h3>gst_rtsp_client_get_mount_points ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
+gst_rtsp_client_get_mount_points (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
+<p>Get the <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> object that <em class="parameter"><code>client</code></em>
+ uses to manage its sessions.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.6.6"></a><h4>Returns</h4>
+<p> a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>, unref after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-set-mount-points"></a><h3>gst_rtsp_client_set_mount_points ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_client_set_mount_points (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>);</pre>
+<p>Set <em class="parameter"><code>mounts</code></em>
+ as the mount points for <em class="parameter"><code>client</code></em>
+ which it will use to map urls
+to media streams. These mount points are usually inherited from the server that
+created the client but can be overriden later.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>mounts</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-get-auth"></a><h3>gst_rtsp_client_get_auth ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
+gst_rtsp_client_get_auth (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
+<p>Get the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> used as the authentication manager of <em class="parameter"><code>client</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.8.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> of <em class="parameter"><code>client</code></em>
+. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-set-auth"></a><h3>gst_rtsp_client_set_auth ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_client_set_auth (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>);</pre>
+<p>configure <em class="parameter"><code>auth</code></em>
+ to be used as the authentication manager of <em class="parameter"><code>client</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>auth</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-get-thread-pool"></a><h3>gst_rtsp_client_get_thread_pool ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
+gst_rtsp_client_get_thread_pool (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
+<p>Get the <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> used as the thread pool of <em class="parameter"><code>client</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.10.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> of <em class="parameter"><code>client</code></em>
+. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-set-thread-pool"></a><h3>gst_rtsp_client_set_thread_pool ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_client_set_thread_pool (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> *pool</code></em>);</pre>
+<p>configure <em class="parameter"><code>pool</code></em>
+ to be used as the thread pool of <em class="parameter"><code>client</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.11.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-get-connection"></a><h3>gst_rtsp_client_get_connection ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="returnvalue">GstRTSPConnection</span></a> *
+gst_rtsp_client_get_connection (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
+<p>Get the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> of <em class="parameter"><code>client</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.12.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.12.6"></a><h4>Returns</h4>
+<p> the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> of <em class="parameter"><code>client</code></em>
+.
+The connection object returned remains valid until the client is freed. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-set-connection"></a><h3>gst_rtsp_client_set_connection ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_client_set_connection (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> *conn</code></em>);</pre>
+<p>Set the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> of <em class="parameter"><code>client</code></em>
+. This function takes ownership of
+<em class="parameter"><code>conn</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.13.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>conn</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.13.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-attach"></a><h3>gst_rtsp_client_attach ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_client_attach (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a> *context</code></em>);</pre>
+<p>Attaches <em class="parameter"><code>client</code></em>
+ to <em class="parameter"><code>context</code></em>
+. When the mainloop for <em class="parameter"><code>context</code></em>
+ is run, the
+client will be dispatched. When <em class="parameter"><code>context</code></em>
+ is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, the default context will be
+used).</p>
+<p>This function should be called when the client properties and urls are fully
+configured and the client is ready to start.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.14.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>context</p></td>
+<td class="parameter_description"><p> a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.14.7"></a><h4>Returns</h4>
+<p> the ID (greater than 0) for the source within the GMainContext.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClientSendFunc"></a><h3>GstRTSPClientSendFunc ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+<span class="c_punctuation">(</span>*GstRTSPClientSendFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> *message</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> close</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>This callback is called when <em class="parameter"><code>client</code></em>
+ wants to send <em class="parameter"><code>message</code></em>
+. When <em class="parameter"><code>close</code></em>
+ is
+<a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a>, the connection should be closed when the message has been sent.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.15.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>message</p></td>
+<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>close</p></td>
+<td class="parameter_description"><p>close the connection</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data when registering the callback</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.15.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-set-send-func"></a><h3>gst_rtsp_client_set_send_func ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_client_set_send_func (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPClient.html#GstRTSPClientSendFunc" title="GstRTSPClientSendFunc ()"><span class="type">GstRTSPClientSendFunc</span></a> func</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Datasets.html#GDestroyNotify"><span class="type">GDestroyNotify</span></a> notify</code></em>);</pre>
+<p>Set <em class="parameter"><code>func</code></em>
+ as the callback that will be called when a new message needs to be
+sent to the client. <em class="parameter"><code>user_data</code></em>
+ is passed to <em class="parameter"><code>func</code></em>
+ and <em class="parameter"><code>notify</code></em>
+ is called when
+<em class="parameter"><code>user_data</code></em>
+ is no longer in use.</p>
+<p>By default, the client will send the messages on the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> that
+was configured with <a class="link" href="GstRTSPClient.html#gst-rtsp-client-attach" title="gst_rtsp_client_attach ()"><code class="function">gst_rtsp_client_attach()</code></a> was called.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.16.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>func</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPClient.html#GstRTSPClientSendFunc" title="GstRTSPClientSendFunc ()"><span class="type">GstRTSPClientSendFunc</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid until the GDestroyNotify argument is called."><span class="acronym">scope notified</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p> user data passed to <em class="parameter"><code>func</code></em>
+. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>notify</p></td>
+<td class="parameter_description"><p> called when <em class="parameter"><code>user_data</code></em>
+is no longer in use. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-handle-message"></a><h3>gst_rtsp_client_handle_message ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
+gst_rtsp_client_handle_message (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> *message</code></em>);</pre>
+<p>Let the client handle <em class="parameter"><code>message</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.17.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>message</p></td>
+<td class="parameter_description"><p> an <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.17.6"></a><h4>Returns</h4>
+<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="type">GstRTSPResult</span></a>.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-send-message"></a><h3>gst_rtsp_client_send_message ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
+gst_rtsp_client_send_message (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> *message</code></em>);</pre>
+<p>Send a message message to the remote end. <em class="parameter"><code>message</code></em>
+ must be a
+<span class="type">GST_RTSP_MESSAGE_REQUEST</span> or a <span class="type">GST_RTSP_MESSAGE_RESPONSE</span>.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.18.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> to send
+the message to or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>][<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>message</p></td>
+<td class="parameter_description"><p> The <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> to send. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClientSessionFilterFunc"></a><h3>GstRTSPClientSessionFilterFunc ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
+<span class="c_punctuation">(</span>*GstRTSPClientSessionFilterFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>This function will be called by the <a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()"><code class="function">gst_rtsp_client_session_filter()</code></a>. An
+implementation should return a value of <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
+<p>When this function returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, <em class="parameter"><code>sess</code></em>
+ will be removed
+from <em class="parameter"><code>client</code></em>
+.</p>
+<p>A return value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a> will leave <em class="parameter"><code>sess</code></em>
+ untouched in
+<em class="parameter"><code>client</code></em>
+.</p>
+<p>A value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will add <em class="parameter"><code>sess</code></em>
+ to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()"><code class="function">gst_rtsp_client_session_filter()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.19.8"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> object</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>sess</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> in <em class="parameter"><code>client</code></em>
+</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data that has been given to <a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()"><code class="function">gst_rtsp_client_session_filter()</code></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.19.9"></a><h4>Returns</h4>
+<p> a <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-client-session-filter"></a><h3>gst_rtsp_client_session_filter ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
+gst_rtsp_client_session_filter (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPClient.html#GstRTSPClientSessionFilterFunc" title="GstRTSPClientSessionFilterFunc ()"><span class="type">GstRTSPClientSessionFilterFunc</span></a> func</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>Call <em class="parameter"><code>func</code></em>
+ for each session managed by <em class="parameter"><code>client</code></em>
+. The result value of <em class="parameter"><code>func</code></em>
+
+determines what happens to the session. <em class="parameter"><code>func</code></em>
+ will be called with <em class="parameter"><code>client</code></em>
+
+locked so no further actions on <em class="parameter"><code>client</code></em>
+ can be performed from <em class="parameter"><code>func</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, the session will be removed from
+<em class="parameter"><code>client</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a>, the session will remain in <em class="parameter"><code>client</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>, the session will remain in <em class="parameter"><code>client</code></em>
+ but
+will also be added with an additional ref to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of this
+function..</p>
+<p>When <em class="parameter"><code>func</code></em>
+ is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will be assumed for each session.</p>
+<div class="refsect3">
+<a name="id-1.2.2.9.20.9"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>func</p></td>
+<td class="parameter_description"><p> a callback. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid only during the call to the method."><span class="acronym">scope call</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data passed to <em class="parameter"><code>func</code></em>
+</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.2.9.20.10"></a><h4>Returns</h4>
+<p> a <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> with all
+sessions for which <em class="parameter"><code>func</code></em>
+returned <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>. After usage, each
+element in the <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> should be unreffed before the list is freed. </p>
+<p><span class="annotation">[<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRTSPSession][<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPClient.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPClient-struct"></a><h3>struct GstRTSPClient</h3>
+<pre class="programlisting">struct GstRTSPClient;</pre>
+<p>The client object represents the connection and its state with a client.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClientClass"></a><h3>struct GstRTSPClientClass</h3>
+<pre class="programlisting">struct GstRTSPClientClass {
+ GObjectClass parent_class;
+
+ GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
+ gboolean (*configure_client_media) (GstRTSPClient * client,
+ GstRTSPMedia * media, GstRTSPStream * stream,
+ GstRTSPContext * ctx);
+ gboolean (*configure_client_transport) (GstRTSPClient * client,
+ GstRTSPContext * ctx,
+ GstRTSPTransport * ct);
+ GstRTSPResult (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPResult (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
+ gchar * (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri);
+
+ /* signals */
+ void (*closed) (GstRTSPClient *client);
+ void (*new_session) (GstRTSPClient *client, GstRTSPSession *session);
+ void (*options_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*describe_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*setup_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*play_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*pause_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*handle_response) (GstRTSPClient *client, GstRTSPContext *ctx);
+
+ void (*tunnel_http_response) (GstRTSPClient * client, GstRTSPMessage * request,
+ GstRTSPMessage * response);
+};
+</pre>
+<p>The client class structure.</p>
+<div class="refsect3">
+<a name="id-1.2.2.10.3.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPClientClass.parent-class"></a>parent_class</code></em>;</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.create-sdp"></a>create_sdp</code></em> ()</p></td>
+<td class="struct_member_description"><p>called when the SDP needs to be created for media.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.configure-client-media"></a>configure_client_media</code></em> ()</p></td>
+<td class="struct_member_description"><p>called when the stream in media needs to be configured.
+The default implementation will configure the blocksize on the payloader when
+spcified in the request headers.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.configure-client-transport"></a>configure_client_transport</code></em> ()</p></td>
+<td class="struct_member_description"><p>called when the client transport needs to be
+configured.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.params-set"></a>params_set</code></em> ()</p></td>
+<td class="struct_member_description"><p>set parameters. This function should also initialize the
+RTSP response(ctx->response) via a call to <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#gst-rtsp-message-init-response"><code class="function">gst_rtsp_message_init_response()</code></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.params-get"></a>params_get</code></em> ()</p></td>
+<td class="struct_member_description"><p>get parameters. This function should also initialize the
+RTSP response(ctx->response) via a call to <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#gst-rtsp-message-init-response"><code class="function">gst_rtsp_message_init_response()</code></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.make-path-from-uri"></a>make_path_from_uri</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.closed"></a>closed</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.new-session"></a>new_session</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.options-request"></a>options_request</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.describe-request"></a>describe_request</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.setup-request"></a>setup_request</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.play-request"></a>play_request</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.pause-request"></a>pause_request</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.teardown-request"></a>teardown_request</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.set-parameter-request"></a>set_parameter_request</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.get-parameter-request"></a>get_parameter_request</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.handle-response"></a>handle_response</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.tunnel-http-response"></a>tunnel_http_response</code></em> ()</p></td>
+<td class="struct_member_description"><p>called when a response to the GET request is about to
+be sent for a tunneled connection. The response can be modified. Since 1.4</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPClient.property-details"></a><h2>Property Details</h2>
+<div class="refsect2">
+<a name="GstRTSPClient--drop-backlog"></a><h3>The <code class="literal">“drop-backlog”</code> property</h3>
+<pre class="programlisting"> “drop-backlog” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
+<p>Drop data when the backlog queue is full.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: TRUE</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient--mount-points"></a><h3>The <code class="literal">“mount-points”</code> property</h3>
+<pre class="programlisting"> “mount-points” <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *</pre>
+<p>The mount points to use for client session.</p>
+<p>Flags: Read / Write</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient--session-pool"></a><h3>The <code class="literal">“session-pool”</code> property</h3>
+<pre class="programlisting"> “session-pool” <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *</pre>
+<p>The session pool to use for client session.</p>
+<p>Flags: Read / Write</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPClient.signal-details"></a><h2>Signal Details</h2>
+<div class="refsect2">
+<a name="GstRTSPClient-closed"></a><h3>The <code class="literal">“closed”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-describe-request"></a><h3>The <code class="literal">“describe-request”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-get-parameter-request"></a><h3>The <code class="literal">“get-parameter-request”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-handle-response"></a><h3>The <code class="literal">“handle-response”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-new-session"></a><h3>The <code class="literal">“new-session”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-options-request"></a><h3>The <code class="literal">“options-request”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-pause-request"></a><h3>The <code class="literal">“pause-request”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-play-request"></a><h3>The <code class="literal">“play-request”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-send-message"></a><h3>The <code class="literal">“send-message”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client,
+ <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *session,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> message,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<div class="refsect3">
+<a name="id-1.2.2.12.10.4"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>The RTSP client</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p> The session. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Override the parsed C type with given type."><span class="acronym">type</span></acronym> GstRtspServer.RTSPSession]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>message</p></td>
+<td class="parameter_description"><p> The message. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Override the parsed C type with given type."><span class="acronym">type</span></acronym> GstRtsp.RTSPMessage]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data set when the signal handler was connected.</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-set-parameter-request"></a><h3>The <code class="literal">“set-parameter-request”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-setup-request"></a><h3>The <code class="literal">“setup-request”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPClient-teardown-request"></a><h3>The <code class="literal">“teardown-request”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
+ <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPClient.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
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+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
+<html>
+<head>
+<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
+<title>GStreamer RTSP Server Reference Manual: GstRTSPMedia</title>
+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
+<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
+<link rel="up" href="ch01.html" title="">
+<link rel="prev" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">
+<link rel="next" href="GstRTSPStream.html" title="GstRTSPStream">
+<meta name="generator" content="GTK-Doc V1.21 (XML mode)">
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+<table class="navigation" id="top" width="100%" summary="Navigation header" cellpadding="2" cellspacing="5"><tr valign="middle">
+<td width="100%" align="left" class="shortcuts">
+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#GstRTSPMedia.description" class="shortcut">Description</a></span><span id="nav_hierarchy"> <span class="dim">|</span>
+ <a href="#GstRTSPMedia.object-hierarchy" class="shortcut">Object Hierarchy</a></span><span id="nav_properties"> <span class="dim">|</span>
+ <a href="#GstRTSPMedia.properties" class="shortcut">Properties</a></span><span id="nav_signals"> <span class="dim">|</span>
+ <a href="#GstRTSPMedia.signals" class="shortcut">Signals</a></span>
+</td>
+<td><a accesskey="h" href="index.html"><img src="home.png" width="16" height="16" border="0" alt="Home"></a></td>
+<td><a accesskey="u" href="ch01.html"><img src="up.png" width="16" height="16" border="0" alt="Up"></a></td>
+<td><a accesskey="p" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html"><img src="left.png" width="16" height="16" border="0" alt="Prev"></a></td>
+<td><a accesskey="n" href="GstRTSPStream.html"><img src="right.png" width="16" height="16" border="0" alt="Next"></a></td>
+</tr></table>
+<div class="refentry">
+<a name="GstRTSPMedia"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="GstRTSPMedia.top_of_page"></a>GstRTSPMedia</span></h2>
+<p>GstRTSPMedia — The media pipeline</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="GstRTSPMedia.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-new" title="gst_rtsp_media_new ()">gst_rtsp_media_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="returnvalue">GstElement</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-element" title="gst_rtsp_media_get_element ()">gst_rtsp_media_get_element</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-take-pipeline" title="gst_rtsp_media_take_pipeline ()">gst_rtsp_media_take_pipeline</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-permissions" title="gst_rtsp_media_set_permissions ()">gst_rtsp_media_set_permissions</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-permissions" title="gst_rtsp_media_get_permissions ()">gst_rtsp_media_get_permissions</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-shared" title="gst_rtsp_media_set_shared ()">gst_rtsp_media_set_shared</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-shared" title="gst_rtsp_media_is_shared ()">gst_rtsp_media_is_shared</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-reusable" title="gst_rtsp_media_set_reusable ()">gst_rtsp_media_set_reusable</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-reusable" title="gst_rtsp_media_is_reusable ()">gst_rtsp_media_is_reusable</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-profiles" title="gst_rtsp_media_set_profiles ()">gst_rtsp_media_set_profiles</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-profiles" title="gst_rtsp_media_get_profiles ()">gst_rtsp_media_get_profiles</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-protocols" title="gst_rtsp_media_set_protocols ()">gst_rtsp_media_set_protocols</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-protocols" title="gst_rtsp_media_get_protocols ()">gst_rtsp_media_get_protocols</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-eos-shutdown" title="gst_rtsp_media_set_eos_shutdown ()">gst_rtsp_media_set_eos_shutdown</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-eos-shutdown" title="gst_rtsp_media_is_eos_shutdown ()">gst_rtsp_media_is_eos_shutdown</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-address-pool" title="gst_rtsp_media_set_address_pool ()">gst_rtsp_media_set_address_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-address-pool" title="gst_rtsp_media_get_address_pool ()">gst_rtsp_media_get_address_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-buffer-size" title="gst_rtsp_media_set_buffer_size ()">gst_rtsp_media_set_buffer_size</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-buffer-size" title="gst_rtsp_media_get_buffer_size ()">gst_rtsp_media_get_buffer_size</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-setup-sdp" title="gst_rtsp_media_setup_sdp ()">gst_rtsp_media_setup_sdp</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()">gst_rtsp_media_prepare</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unprepare" title="gst_rtsp_media_unprepare ()">gst_rtsp_media_unprepare</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMedia.html#GstRTSPMediaStatus" title="enum GstRTSPMediaStatus"><span class="returnvalue">GstRTSPMediaStatus</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-status" title="gst_rtsp_media_get_status ()">gst_rtsp_media_get_status</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-suspend-mode" title="gst_rtsp_media_set_suspend_mode ()">gst_rtsp_media_set_suspend_mode</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="returnvalue">GstRTSPSuspendMode</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-suspend-mode" title="gst_rtsp_media_get_suspend_mode ()">gst_rtsp_media_get_suspend_mode</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-suspend" title="gst_rtsp_media_suspend ()">gst_rtsp_media_suspend</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unsuspend" title="gst_rtsp_media_unsuspend ()">gst_rtsp_media_unsuspend</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-collect-streams" title="gst_rtsp_media_collect_streams ()">gst_rtsp_media_collect_streams</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-create-stream" title="gst_rtsp_media_create_stream ()">gst_rtsp_media_create_stream</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-n-streams" title="gst_rtsp_media_n_streams ()">gst_rtsp_media_n_streams</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-stream" title="gst_rtsp_media_get_stream ()">gst_rtsp_media_get_stream</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-find-stream" title="gst_rtsp_media_find_stream ()">gst_rtsp_media_find_stream</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-seek" title="gst_rtsp_media_seek ()">gst_rtsp_media_seek</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-range-string" title="gst_rtsp_media_get_range_string ()">gst_rtsp_media_get_range_string</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-state" title="gst_rtsp_media_set_state ()">gst_rtsp_media_set_state</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-pipeline-state" title="gst_rtsp_media_set_pipeline_state ()">gst_rtsp_media_set_pipeline_state</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html"><span class="returnvalue">GstClock</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-clock" title="gst_rtsp_media_get_clock ()">gst_rtsp_media_get_clock</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="returnvalue">GstClockTime</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-base-time" title="gst_rtsp_media_get_base_time ()">gst_rtsp_media_get_base_time</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-use-time-provider" title="gst_rtsp_media_use_time_provider ()">gst_rtsp_media_use_time_provider</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-time-provider" title="gst_rtsp_media_is_time_provider ()">gst_rtsp_media_is_time_provider</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="returnvalue">GstNetTimeProvider</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-time-provider" title="gst_rtsp_media_get_time_provider ()">gst_rtsp_media_get_time_provider</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMedia.properties"></a><h2>Properties</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="properties_type">
+<col width="300px" class="properties_name">
+<col width="200px" class="properties_flags">
+</colgroup>
+<tbody>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--buffer-size" title="The “buffer-size” property">buffer-size</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *</td>
+<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--element" title="The “element” property">element</a></td>
+<td class="property_flags">Read / Write / Construct Only</td>
+</tr>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--eos-shutdown" title="The “eos-shutdown” property">eos-shutdown</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--profiles" title="The “profiles” property">profiles</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--protocols" title="The “protocols” property">protocols</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--reusable" title="The “reusable” property">reusable</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--shared" title="The “shared” property">shared</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--suspend-mode" title="The “suspend-mode” property">suspend-mode</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--time-provider" title="The “time-provider” property">time-provider</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMedia.signals"></a><h2>Signals</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="signals_return">
+<col width="300px" class="signals_name">
+<col width="200px" class="signals_flags">
+</colgroup>
+<tbody>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-new-state" title="The “new-state” signal">new-state</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-new-stream" title="The “new-stream” signal">new-stream</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-prepared" title="The “prepared” signal">prepared</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-removed-stream" title="The “removed-stream” signal">removed-stream</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-target-state" title="The “target-state” signal">target-state</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-unprepared" title="The “unprepared” signal">unprepared</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMedia.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-struct" title="struct GstRTSPMedia">GstRTSPMedia</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMediaClass" title="struct GstRTSPMediaClass">GstRTSPMediaClass</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">enum</td>
+<td class="function_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMediaStatus" title="enum GstRTSPMediaStatus">GstRTSPMediaStatus</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">enum</td>
+<td class="function_name"><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode">GstRTSPSuspendMode</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMedia.object-hierarchy"></a><h2>Object Hierarchy</h2>
+<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+ <span class="lineart">╰──</span> GstRTSPMedia
+</pre>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMedia.description"></a><h2>Description</h2>
+<p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> contains the complete GStreamer pipeline to manage the
+streaming to the clients. The actual data transfer is done by the
+<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> objects that are created and exposed by the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>.</p>
+<p>The <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> is usually created from a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> when the
+client does a DESCRIBE or SETUP of a resource.</p>
+<p>A media is created with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-new" title="gst_rtsp_media_new ()"><code class="function">gst_rtsp_media_new()</code></a> that takes the element that will
+provide the streaming elements. For each of the streams, a new <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>
+object needs to be made with the <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-create-stream" title="gst_rtsp_media_create_stream ()"><code class="function">gst_rtsp_media_create_stream()</code></a> which takes
+the payloader element and the source pad that produces the RTP stream.</p>
+<p>The pipeline of the media is set to PAUSED with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>. The
+prepare method will add rtpbin and sinks and sources to send and receive RTP
+and RTCP packets from the clients. Each stream srcpad is connected to an
+input into the internal rtpbin.</p>
+<p>It is also possible to dynamically create <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> objects during the
+prepare phase. With <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-status" title="gst_rtsp_media_get_status ()"><code class="function">gst_rtsp_media_get_status()</code></a> you can check the status of
+the prepare phase.</p>
+<p>After the media is prepared, it is ready for streaming. It will usually be
+managed in a session with <a class="link" href="GstRTSPSession.html#gst-rtsp-session-manage-media" title="gst_rtsp_session_manage_media ()"><code class="function">gst_rtsp_session_manage_media()</code></a>. See
+<a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> and <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a>.</p>
+<p>The state of the media can be controlled with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-state" title="gst_rtsp_media_set_state ()"><code class="function">gst_rtsp_media_set_state()</code></a>.
+Seeking can be done with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-seek" title="gst_rtsp_media_seek ()"><code class="function">gst_rtsp_media_seek()</code></a>.</p>
+<p>With <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unprepare" title="gst_rtsp_media_unprepare ()"><code class="function">gst_rtsp_media_unprepare()</code></a> the pipeline is stopped and shut down. When
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-eos-shutdown" title="gst_rtsp_media_set_eos_shutdown ()"><code class="function">gst_rtsp_media_set_eos_shutdown()</code></a> an EOS will be sent to the pipeline to
+cleanly shut down.</p>
+<p>With <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-shared" title="gst_rtsp_media_set_shared ()"><code class="function">gst_rtsp_media_set_shared()</code></a>, the media can be shared between multiple
+clients. With <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-reusable" title="gst_rtsp_media_set_reusable ()"><code class="function">gst_rtsp_media_set_reusable()</code></a> you can control if the pipeline
+can be prepared again after an unprepare.</p>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMedia.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-media-new"></a><h3>gst_rtsp_media_new ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
+gst_rtsp_media_new (<em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *element</code></em>);</pre>
+<p>Create a new <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> instance. <em class="parameter"><code>element</code></em>
+ is the bin element that
+provides the different streams. The <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> object contains the
+element to produce RTP data for one or more related (audio/video/..)
+streams.</p>
+<p>Ownership is taken of <em class="parameter"><code>element</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.2.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>element</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.2.7"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> object. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-element"></a><h3>gst_rtsp_media_get_element ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="returnvalue">GstElement</span></a> *
+gst_rtsp_media_get_element (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get the element that was used when constructing <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.3.6"></a><h4>Returns</h4>
+<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a>. Unref after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-take-pipeline"></a><h3>gst_rtsp_media_take_pipeline ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_take_pipeline (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPipeline.html"><span class="type">GstPipeline</span></a> *pipeline</code></em>);</pre>
+<p>Set <em class="parameter"><code>pipeline</code></em>
+ as the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPipeline.html"><span class="type">GstPipeline</span></a> for <em class="parameter"><code>media</code></em>
+. Ownership is
+taken of <em class="parameter"><code>pipeline</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>pipeline</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPipeline.html"><span class="type">GstPipeline</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-permissions"></a><h3>gst_rtsp_media_set_permissions ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_set_permissions (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>);</pre>
+<p>Set <em class="parameter"><code>permissions</code></em>
+ on <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>permissions</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-permissions"></a><h3>gst_rtsp_media_get_permissions ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
+gst_rtsp_media_get_permissions (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get the permissions object from <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.6.6"></a><h4>Returns</h4>
+<p> a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> object, unref after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-shared"></a><h3>gst_rtsp_media_set_shared ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_set_shared (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> shared</code></em>);</pre>
+<p>Set or unset if the pipeline for <em class="parameter"><code>media</code></em>
+ can be shared will multiple clients.
+When <em class="parameter"><code>shared</code></em>
+ is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a>, client requests for this media will share the media
+pipeline.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>shared</p></td>
+<td class="parameter_description"><p>the new value</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-is-shared"></a><h3>gst_rtsp_media_is_shared ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_is_shared (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Check if the pipeline for <em class="parameter"><code>media</code></em>
+ can be shared between multiple clients.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.8.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media can be shared between clients.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-reusable"></a><h3>gst_rtsp_media_set_reusable ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_set_reusable (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> reusable</code></em>);</pre>
+<p>Set or unset if the pipeline for <em class="parameter"><code>media</code></em>
+ can be reused after the pipeline has
+been unprepared.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>reusable</p></td>
+<td class="parameter_description"><p>the new value</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-is-reusable"></a><h3>gst_rtsp_media_is_reusable ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_is_reusable (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Check if the pipeline for <em class="parameter"><code>media</code></em>
+ can be reused after an unprepare.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.10.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media can be reused</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-profiles"></a><h3>gst_rtsp_media_set_profiles ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_set_profiles (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a> profiles</code></em>);</pre>
+<p>Configure the allowed lower transport for <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.11.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>profiles</p></td>
+<td class="parameter_description"><p>the new flags</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-profiles"></a><h3>gst_rtsp_media_get_profiles ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
+gst_rtsp_media_get_profiles (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get the allowed profiles of <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.12.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.12.6"></a><h4>Returns</h4>
+<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-protocols"></a><h3>gst_rtsp_media_set_protocols ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_set_protocols (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a> protocols</code></em>);</pre>
+<p>Configure the allowed lower transport for <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.13.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>protocols</p></td>
+<td class="parameter_description"><p>the new flags</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-protocols"></a><h3>gst_rtsp_media_get_protocols ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
+gst_rtsp_media_get_protocols (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get the allowed protocols of <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.14.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.14.6"></a><h4>Returns</h4>
+<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-eos-shutdown"></a><h3>gst_rtsp_media_set_eos_shutdown ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_set_eos_shutdown (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> eos_shutdown</code></em>);</pre>
+<p>Set or unset if an EOS event will be sent to the pipeline for <em class="parameter"><code>media</code></em>
+ before
+it is unprepared.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.15.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>eos_shutdown</p></td>
+<td class="parameter_description"><p>the new value</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-is-eos-shutdown"></a><h3>gst_rtsp_media_is_eos_shutdown ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_is_eos_shutdown (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Check if the pipeline for <em class="parameter"><code>media</code></em>
+ will send an EOS down the pipeline before
+unpreparing.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.16.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.16.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media will send EOS before unpreparing.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-address-pool"></a><h3>gst_rtsp_media_set_address_pool ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_set_address_pool (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
+<p>configure <em class="parameter"><code>pool</code></em>
+ to be used as the address pool of <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.17.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-address-pool"></a><h3>gst_rtsp_media_get_address_pool ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
+gst_rtsp_media_get_address_pool (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> used as the address pool of <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.18.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.18.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> of <em class="parameter"><code>media</code></em>
+. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-buffer-size"></a><h3>gst_rtsp_media_set_buffer_size ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_set_buffer_size (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> size</code></em>);</pre>
+<p>Set the kernel UDP buffer size.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.19.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>size</p></td>
+<td class="parameter_description"><p>the new value</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-buffer-size"></a><h3>gst_rtsp_media_get_buffer_size ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_media_get_buffer_size (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get the kernel UDP buffer size.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.20.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.20.6"></a><h4>Returns</h4>
+<p> the kernel UDP buffer size.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-setup-sdp"></a><h3>gst_rtsp_media_setup_sdp ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_setup_sdp (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstsdpmessage.html#GstSDPMessage"><span class="type">GstSDPMessage</span></a> *sdp</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo"><span class="type">GstSDPInfo</span></a> *info</code></em>);</pre>
+<p>Add <em class="parameter"><code>media</code></em>
+ specific info to <em class="parameter"><code>sdp</code></em>
+. <em class="parameter"><code>info</code></em>
+ is used to configure the connection
+information in the SDP.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.21.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>sdp</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstsdpmessage.html#GstSDPMessage"><span class="type">GstSDPMessage</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>info</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo"><span class="type">GstSDPInfo</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.21.6"></a><h4>Returns</h4>
+<p> TRUE on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-prepare"></a><h3>gst_rtsp_media_prepare ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_prepare (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> *thread</code></em>);</pre>
+<p>Prepare <em class="parameter"><code>media</code></em>
+ for streaming. This function will create the objects
+to manage the streaming. A pipeline must have been set on <em class="parameter"><code>media</code></em>
+ with
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-take-pipeline" title="gst_rtsp_media_take_pipeline ()"><code class="function">gst_rtsp_media_take_pipeline()</code></a>.</p>
+<p>It will preroll the pipeline and collect vital information about the streams
+such as the duration.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.22.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>thread</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> to run the
+bus handler or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.22.7"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-unprepare"></a><h3>gst_rtsp_media_unprepare ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_unprepare (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Unprepare <em class="parameter"><code>media</code></em>
+. After this call, the media should be prepared again before
+it can be used again. If the media is set to be non-reusable, a new instance
+must be created.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.23.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.23.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-status"></a><h3>gst_rtsp_media_get_status ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMedia.html#GstRTSPMediaStatus" title="enum GstRTSPMediaStatus"><span class="returnvalue">GstRTSPMediaStatus</span></a>
+gst_rtsp_media_get_status (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get the status of <em class="parameter"><code>media</code></em>
+. When <em class="parameter"><code>media</code></em>
+ is busy preparing, this function waits
+until <em class="parameter"><code>media</code></em>
+ is prepared or in error.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.24.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.24.6"></a><h4>Returns</h4>
+<p> the status of <em class="parameter"><code>media</code></em>
+.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-suspend-mode"></a><h3>gst_rtsp_media_set_suspend_mode ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_set_suspend_mode (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a> mode</code></em>);</pre>
+<p>Control how @ media will be suspended after the SDP has been generated and
+after a PAUSE request has been performed.</p>
+<p>Media must be unprepared when setting the suspend mode.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.25.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>mode</p></td>
+<td class="parameter_description"><p>the new <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-suspend-mode"></a><h3>gst_rtsp_media_get_suspend_mode ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="returnvalue">GstRTSPSuspendMode</span></a>
+gst_rtsp_media_get_suspend_mode (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get how <em class="parameter"><code>media</code></em>
+ will be suspended.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.26.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.26.6"></a><h4>Returns</h4>
+<p> <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a>.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-suspend"></a><h3>gst_rtsp_media_suspend ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_suspend (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Suspend <em class="parameter"><code>media</code></em>
+. The state of the pipeline managed by <em class="parameter"><code>media</code></em>
+ is set to
+GST_STATE_NULL but all streams are kept. <em class="parameter"><code>media</code></em>
+ can be prepared again
+with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unsuspend" title="gst_rtsp_media_unsuspend ()"><code class="function">gst_rtsp_media_unsuspend()</code></a></p>
+<p><em class="parameter"><code>media</code></em>
+ must be prepared with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>;</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.27.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.27.7"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-unsuspend"></a><h3>gst_rtsp_media_unsuspend ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_unsuspend (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Unsuspend <em class="parameter"><code>media</code></em>
+ if it was in a suspended state. This method does nothing
+when the media was not in the suspended state.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.28.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.28.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-collect-streams"></a><h3>gst_rtsp_media_collect_streams ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_collect_streams (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Find all payloader elements, they should be named pay%d in the
+element of <em class="parameter"><code>media</code></em>
+, and create <a href="../gst-rtsp-server-1.0/GstRTSPStream.html"><span class="type">GstRTSPStreams</span></a> for them.</p>
+<p>Collect all dynamic elements, named dynpay%d, and add them to
+the list of dynamic elements.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.29.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-create-stream"></a><h3>gst_rtsp_media_create_stream ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
+gst_rtsp_media_create_stream (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *payloader</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="type">GstPad</span></a> *srcpad</code></em>);</pre>
+<p>Create a new stream in <em class="parameter"><code>media</code></em>
+ that provides RTP data on <em class="parameter"><code>srcpad</code></em>
+.
+<em class="parameter"><code>srcpad</code></em>
+ should be a pad of an element inside <em class="parameter"><code>media->element</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.30.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>payloader</p></td>
+<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>srcpad</p></td>
+<td class="parameter_description"><p>a source <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="type">GstPad</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.30.6"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> that remains valid for as long
+as <em class="parameter"><code>media</code></em>
+exists. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-n-streams"></a><h3>gst_rtsp_media_n_streams ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_media_n_streams (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get the number of streams in this media.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.31.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.31.6"></a><h4>Returns</h4>
+<p> The number of streams.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-stream"></a><h3>gst_rtsp_media_get_stream ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
+gst_rtsp_media_get_stream (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> idx</code></em>);</pre>
+<p>Retrieve the stream with index <em class="parameter"><code>idx</code></em>
+ from <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.32.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>idx</p></td>
+<td class="parameter_description"><p>the stream index</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.32.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> at index
+<em class="parameter"><code>idx</code></em>
+or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when a stream with that index did not exist. </p>
+<p><span class="annotation">[<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>][<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-find-stream"></a><h3>gst_rtsp_media_find_stream ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
+gst_rtsp_media_find_stream (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *control</code></em>);</pre>
+<p>Find a stream in <em class="parameter"><code>media</code></em>
+ with <em class="parameter"><code>control</code></em>
+ as the control uri.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.33.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>control</p></td>
+<td class="parameter_description"><p>the control of the stream</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.33.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> with
+control uri <em class="parameter"><code>control</code></em>
+or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when a stream with that control did
+not exist. </p>
+<p><span class="annotation">[<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>][<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-seek"></a><h3>gst_rtsp_media_seek ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_seek (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsprange.html#GstRTSPTimeRange"><span class="type">GstRTSPTimeRange</span></a> *range</code></em>);</pre>
+<p>Seek the pipeline of <em class="parameter"><code>media</code></em>
+ to <em class="parameter"><code>range</code></em>
+. <em class="parameter"><code>media</code></em>
+ must be prepared with
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.34.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>range</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsprange.html#GstRTSPTimeRange"><span class="type">GstRTSPTimeRange</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.34.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-range-string"></a><h3>gst_rtsp_media_get_range_string ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_media_get_range_string (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> play</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsprange.html#GstRTSPRangeUnit"><span class="type">GstRTSPRangeUnit</span></a> unit</code></em>);</pre>
+<p>Get the current range as a string. <em class="parameter"><code>media</code></em>
+ must be prepared with
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.35.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>play</p></td>
+<td class="parameter_description"><p>for the PLAY request</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>unit</p></td>
+<td class="parameter_description"><p>the unit to use for the string</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.35.6"></a><h4>Returns</h4>
+<p> The range as a string, <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-state"></a><h3>gst_rtsp_media_set_state ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_set_state (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GstState"><span class="type">GstState</span></a> state</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Pointer-Arrays.html#GPtrArray"><span class="type">GPtrArray</span></a> *transports</code></em>);</pre>
+<p>Set the state of <em class="parameter"><code>media</code></em>
+ to <em class="parameter"><code>state</code></em>
+ and for the transports in <em class="parameter"><code>transports</code></em>
+.</p>
+<p><em class="parameter"><code>media</code></em>
+ must be prepared with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>;</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.36.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>state</p></td>
+<td class="parameter_description"><p>the target state of the media</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>transports</p></td>
+<td class="parameter_description"><p>a <a href="https://developer.gnome.org/glib/unstable/glib-Pointer-Arrays.html#GPtrArray"><span class="type">GPtrArray</span></a> of <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> pointers. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>][<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRtspServer.RTSPStreamTransport]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.36.7"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-set-pipeline-state"></a><h3>gst_rtsp_media_set_pipeline_state ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_set_pipeline_state (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GstState"><span class="type">GstState</span></a> state</code></em>);</pre>
+<p>Set the state of the pipeline managed by <em class="parameter"><code>media</code></em>
+ to <em class="parameter"><code>state</code></em>
+</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.37.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>state</p></td>
+<td class="parameter_description"><p>the target state of the pipeline</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-clock"></a><h3>gst_rtsp_media_get_clock ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html"><span class="returnvalue">GstClock</span></a> *
+gst_rtsp_media_get_clock (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get the clock that is used by the pipeline in <em class="parameter"><code>media</code></em>
+.</p>
+<p><em class="parameter"><code>media</code></em>
+ must be prepared before this method returns a valid clock object.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.38.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.38.7"></a><h4>Returns</h4>
+<p> the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html"><span class="type">GstClock</span></a> used by <em class="parameter"><code>media</code></em>
+. unref after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-base-time"></a><h3>gst_rtsp_media_get_base_time ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="returnvalue">GstClockTime</span></a>
+gst_rtsp_media_get_base_time (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Get the base_time that is used by the pipeline in <em class="parameter"><code>media</code></em>
+.</p>
+<p><em class="parameter"><code>media</code></em>
+ must be prepared before this method returns a valid base_time.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.39.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.39.7"></a><h4>Returns</h4>
+<p> the base_time used by <em class="parameter"><code>media</code></em>
+.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-use-time-provider"></a><h3>gst_rtsp_media_use_time_provider ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_use_time_provider (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> time_provider</code></em>);</pre>
+<p>Set <em class="parameter"><code>media</code></em>
+ to provide a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.40.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>time_provider</p></td>
+<td class="parameter_description"><p>if a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a> should be used</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-is-time-provider"></a><h3>gst_rtsp_media_is_time_provider ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_is_time_provider (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Check if <em class="parameter"><code>media</code></em>
+ can provide a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a> for its pipeline clock.</p>
+<p>Use <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-time-provider" title="gst_rtsp_media_get_time_provider ()"><code class="function">gst_rtsp_media_get_time_provider()</code></a> to get the network clock.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.41.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.41.7"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>media</code></em>
+can provide a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a>.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-get-time-provider"></a><h3>gst_rtsp_media_get_time_provider ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="returnvalue">GstNetTimeProvider</span></a> *
+gst_rtsp_media_get_time_provider (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *address</code></em>,
+ <em class="parameter"><code><span class="type">guint16</span> port</code></em>);</pre>
+<p>Get the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a> for the clock used by <em class="parameter"><code>media</code></em>
+. The time provider
+will listen on <em class="parameter"><code>address</code></em>
+ and <em class="parameter"><code>port</code></em>
+ for client time requests.</p>
+<div class="refsect3">
+<a name="id-1.2.7.9.42.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>address</p></td>
+<td class="parameter_description"><p> an address or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>port</p></td>
+<td class="parameter_description"><p>a port or 0</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.7.9.42.6"></a><h4>Returns</h4>
+<p> the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a> of <em class="parameter"><code>media</code></em>
+. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMedia.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPMedia-struct"></a><h3>struct GstRTSPMedia</h3>
+<pre class="programlisting">struct GstRTSPMedia;</pre>
+<p>A class that contains the GStreamer element along with a list of
+<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> objects that can produce data.</p>
+<p>This object is usually created from a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaClass"></a><h3>struct GstRTSPMediaClass</h3>
+<pre class="programlisting">struct GstRTSPMediaClass {
+ GObjectClass parent_class;
+
+ /* vmethods */
+ gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
+ gboolean (*prepare) (GstRTSPMedia *media, GstRTSPThread *thread);
+ gboolean (*unprepare) (GstRTSPMedia *media);
+ gboolean (*suspend) (GstRTSPMedia *media);
+ gboolean (*unsuspend) (GstRTSPMedia *media);
+ gboolean (*convert_range) (GstRTSPMedia *media, GstRTSPTimeRange *range,
+ GstRTSPRangeUnit unit);
+ gboolean (*query_position) (GstRTSPMedia *media, gint64 *position);
+ gboolean (*query_stop) (GstRTSPMedia *media, gint64 *stop);
+ GstElement * (*create_rtpbin) (GstRTSPMedia *media);
+ gboolean (*setup_rtpbin) (GstRTSPMedia *media, GstElement *rtpbin);
+ gboolean (*setup_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);
+
+ /* signals */
+ void (*new_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
+ void (*removed_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
+
+ void (*prepared) (GstRTSPMedia *media);
+ void (*unprepared) (GstRTSPMedia *media);
+
+ void (*target_state) (GstRTSPMedia *media, GstState state);
+ void (*new_state) (GstRTSPMedia *media, GstState state);
+};
+</pre>
+<p>The RTSP media class</p>
+<div class="refsect3">
+<a name="id-1.2.7.10.3.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPMediaClass.parent-class"></a>parent_class</code></em>;</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.handle-message"></a>handle_message</code></em> ()</p></td>
+<td class="struct_member_description"><p>handle a message</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.prepare"></a>prepare</code></em> ()</p></td>
+<td class="struct_member_description"><p>the default implementation adds all elements and sets the
+pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING
+in case of NO_PREROLL elements).</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.unprepare"></a>unprepare</code></em> ()</p></td>
+<td class="struct_member_description"><p>the default implementation sets the pipeline's state
+to GST_STATE_NULL and removes all elements.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.suspend"></a>suspend</code></em> ()</p></td>
+<td class="struct_member_description"><p>the default implementation sets the pipeline's state to
+GST_STATE_NULL GST_STATE_PAUSED depending on the selected
+suspend mode.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.unsuspend"></a>unsuspend</code></em> ()</p></td>
+<td class="struct_member_description"><p>the default implementation reverts the suspend operation.
+The pipeline will be prerolled again if it's state was
+set to GST_STATE_NULL in suspend.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.convert-range"></a>convert_range</code></em> ()</p></td>
+<td class="struct_member_description"><p>convert a range to the given unit</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.query-position"></a>query_position</code></em> ()</p></td>
+<td class="struct_member_description"><p>query the current position in the pipeline</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.query-stop"></a>query_stop</code></em> ()</p></td>
+<td class="struct_member_description"><p>query when playback will stop</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.create-rtpbin"></a>create_rtpbin</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.setup-rtpbin"></a>setup_rtpbin</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.setup-sdp"></a>setup_sdp</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.new-stream"></a>new_stream</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.removed-stream"></a>removed_stream</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.prepared"></a>prepared</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.unprepared"></a>unprepared</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.target-state"></a>target_state</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.new-state"></a>new_state</code></em> ()</p></td>
+<td> </td>
+<td> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaStatus"></a><h3>enum GstRTSPMediaStatus</h3>
+<p>The state of the media pipeline.</p>
+<div class="refsect3">
+<a name="id-1.2.7.10.4.4"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="enum_members_name">
+<col class="enum_members_description">
+<col width="200px" class="enum_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-UNPREPARED:CAPS"></a>GST_RTSP_MEDIA_STATUS_UNPREPARED</p></td>
+<td class="enum_member_description">
+<p>media pipeline not prerolled</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-UNPREPARING:CAPS"></a>GST_RTSP_MEDIA_STATUS_UNPREPARING</p></td>
+<td class="enum_member_description">
+<p>media pipeline is busy doing a clean
+ shutdown.</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-PREPARING:CAPS"></a>GST_RTSP_MEDIA_STATUS_PREPARING</p></td>
+<td class="enum_member_description">
+<p>media pipeline is prerolling</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-PREPARED:CAPS"></a>GST_RTSP_MEDIA_STATUS_PREPARED</p></td>
+<td class="enum_member_description">
+<p>media pipeline is prerolled</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-SUSPENDED:CAPS"></a>GST_RTSP_MEDIA_STATUS_SUSPENDED</p></td>
+<td class="enum_member_description">
+<p>media is suspended</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-ERROR:CAPS"></a>GST_RTSP_MEDIA_STATUS_ERROR</p></td>
+<td class="enum_member_description">
+<p>media pipeline is in error</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPSuspendMode"></a><h3>enum GstRTSPSuspendMode</h3>
+<p>The suspend mode of the media pipeline. A media pipeline is suspended right
+after creating the SDP and when the client performs a PAUSED request.</p>
+<div class="refsect3">
+<a name="id-1.2.7.10.5.4"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="enum_members_name">
+<col class="enum_members_description">
+<col width="200px" class="enum_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-SUSPEND-MODE-NONE:CAPS"></a>GST_RTSP_SUSPEND_MODE_NONE</p></td>
+<td class="enum_member_description">
+<p>Media is not suspended</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-SUSPEND-MODE-PAUSE:CAPS"></a>GST_RTSP_SUSPEND_MODE_PAUSE</p></td>
+<td class="enum_member_description">
+<p>Media is PAUSED in suspend</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-SUSPEND-MODE-RESET:CAPS"></a>GST_RTSP_SUSPEND_MODE_RESET</p></td>
+<td class="enum_member_description">
+<p>The media is set to NULL when suspended</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMedia.property-details"></a><h2>Property Details</h2>
+<div class="refsect2">
+<a name="GstRTSPMedia--buffer-size"></a><h3>The <code class="literal">“buffer-size”</code> property</h3>
+<pre class="programlisting"> “buffer-size” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
+<p>The kernel UDP buffer size to use.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: 524288</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia--element"></a><h3>The <code class="literal">“element”</code> property</h3>
+<pre class="programlisting"> “element” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *</pre>
+<p>The GstBin to use for streaming the media.</p>
+<p>Flags: Read / Write / Construct Only</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia--eos-shutdown"></a><h3>The <code class="literal">“eos-shutdown”</code> property</h3>
+<pre class="programlisting"> “eos-shutdown” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
+<p>Send an EOS event to the pipeline before unpreparing.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: FALSE</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia--profiles"></a><h3>The <code class="literal">“profiles”</code> property</h3>
+<pre class="programlisting"> “profiles” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></pre>
+<p>Allowed transfer profiles.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: GST_RTSP_PROFILE_AVP</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia--protocols"></a><h3>The <code class="literal">“protocols”</code> property</h3>
+<pre class="programlisting"> “protocols” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></pre>
+<p>Allowed lower transport protocols.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia--reusable"></a><h3>The <code class="literal">“reusable”</code> property</h3>
+<pre class="programlisting"> “reusable” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
+<p>If this media pipeline can be reused after an unprepare.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: FALSE</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia--shared"></a><h3>The <code class="literal">“shared”</code> property</h3>
+<pre class="programlisting"> “shared” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
+<p>If this media pipeline can be shared.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: FALSE</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia--suspend-mode"></a><h3>The <code class="literal">“suspend-mode”</code> property</h3>
+<pre class="programlisting"> “suspend-mode” <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></pre>
+<p>How to suspend the media in PAUSED.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: GST_RTSP_SUSPEND_MODE_NONE</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia--time-provider"></a><h3>The <code class="literal">“time-provider”</code> property</h3>
+<pre class="programlisting"> “time-provider” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
+<p>Use a NetTimeProvider for clients.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: FALSE</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMedia.signal-details"></a><h2>Signal Details</h2>
+<div class="refsect2">
+<a name="GstRTSPMedia-new-state"></a><h3>The <code class="literal">“new-state”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia-new-stream"></a><h3>The <code class="literal">“new-stream”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
+ <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia-prepared"></a><h3>The <code class="literal">“prepared”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia-removed-stream"></a><h3>The <code class="literal">“removed-stream”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
+ <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia-target-state"></a><h3>The <code class="literal">“target-state”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMedia-unprepared"></a><h3>The <code class="literal">“unprepared”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMedia.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>, <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>, <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a>,
+ <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
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+ <a href="#GstRTSPMediaFactory.properties" class="shortcut">Properties</a></span><span id="nav_signals"> <span class="dim">|</span>
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+<div class="refentry">
+<a name="GstRTSPMediaFactory"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="GstRTSPMediaFactory.top_of_page"></a>GstRTSPMediaFactory</span></h2>
+<p>GstRTSPMediaFactory — A factory for media pipelines</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="returnvalue">GstRTSPMediaFactory</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-new" title="gst_rtsp_media_factory_new ()">gst_rtsp_media_factory_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-launch" title="gst_rtsp_media_factory_get_launch ()">gst_rtsp_media_factory_get_launch</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-launch" title="gst_rtsp_media_factory_set_launch ()">gst_rtsp_media_factory_set_launch</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-permissions" title="gst_rtsp_media_factory_get_permissions ()">gst_rtsp_media_factory_get_permissions</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-permissions" title="gst_rtsp_media_factory_set_permissions ()">gst_rtsp_media_factory_set_permissions</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-add-role" title="gst_rtsp_media_factory_add_role ()">gst_rtsp_media_factory_add_role</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-shared" title="gst_rtsp_media_factory_set_shared ()">gst_rtsp_media_factory_set_shared</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-shared" title="gst_rtsp_media_factory_is_shared ()">gst_rtsp_media_factory_is_shared</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-eos-shutdown" title="gst_rtsp_media_factory_is_eos_shutdown ()">gst_rtsp_media_factory_is_eos_shutdown</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-eos-shutdown" title="gst_rtsp_media_factory_set_eos_shutdown ()">gst_rtsp_media_factory_set_eos_shutdown</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-protocols" title="gst_rtsp_media_factory_get_protocols ()">gst_rtsp_media_factory_get_protocols</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-protocols" title="gst_rtsp_media_factory_set_protocols ()">gst_rtsp_media_factory_set_protocols</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-profiles" title="gst_rtsp_media_factory_set_profiles ()">gst_rtsp_media_factory_set_profiles</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-profiles" title="gst_rtsp_media_factory_get_profiles ()">gst_rtsp_media_factory_get_profiles</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-address-pool" title="gst_rtsp_media_factory_get_address_pool ()">gst_rtsp_media_factory_get_address_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-address-pool" title="gst_rtsp_media_factory_set_address_pool ()">gst_rtsp_media_factory_set_address_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-buffer-size" title="gst_rtsp_media_factory_get_buffer_size ()">gst_rtsp_media_factory_get_buffer_size</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-buffer-size" title="gst_rtsp_media_factory_set_buffer_size ()">gst_rtsp_media_factory_set_buffer_size</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="returnvalue">GstRTSPSuspendMode</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-suspend-mode" title="gst_rtsp_media_factory_get_suspend_mode ()">gst_rtsp_media_factory_get_suspend_mode</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-suspend-mode" title="gst_rtsp_media_factory_set_suspend_mode ()">gst_rtsp_media_factory_set_suspend_mode</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-construct" title="gst_rtsp_media_factory_construct ()">gst_rtsp_media_factory_construct</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="returnvalue">GstElement</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-create-element" title="gst_rtsp_media_factory_create_element ()">gst_rtsp_media_factory_create_element</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.properties"></a><h2>Properties</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="properties_type">
+<col width="300px" class="properties_name">
+<col width="200px" class="properties_flags">
+</colgroup>
+<tbody>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--buffer-size" title="The “buffer-size” property">buffer-size</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--eos-shutdown" title="The “eos-shutdown” property">eos-shutdown</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</td>
+<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--launch" title="The “launch” property">launch</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--profiles" title="The “profiles” property">profiles</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--protocols" title="The “protocols” property">protocols</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--shared" title="The “shared” property">shared</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--suspend-mode" title="The “suspend-mode” property">suspend-mode</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.signals"></a><h2>Signals</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="signals_return">
+<col width="300px" class="signals_name">
+<col width="200px" class="signals_flags">
+</colgroup>
+<tbody>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-configure" title="The “media-configure” signal">media-configure</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+<tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-constructed" title="The “media-constructed” signal">media-constructed</a></td>
+<td class="signal_flags">Run Last</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-struct" title="struct GstRTSPMediaFactory">GstRTSPMediaFactory</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactoryClass" title="struct GstRTSPMediaFactoryClass">GstRTSPMediaFactoryClass</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.object-hierarchy"></a><h2>Object Hierarchy</h2>
+<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+ <span class="lineart">╰──</span> GstRTSPMediaFactory
+</pre>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.description"></a><h2>Description</h2>
+<p>The <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> is responsible for creating or recycling
+<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> objects based on the passed URL.</p>
+<p>The default implementation of the object can create <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> objects
+containing a pipeline created from a launch description set with
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-launch" title="gst_rtsp_media_factory_set_launch ()"><code class="function">gst_rtsp_media_factory_set_launch()</code></a>.</p>
+<p>Media from a factory can be shared by setting the shared flag with
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-shared" title="gst_rtsp_media_factory_set_shared ()"><code class="function">gst_rtsp_media_factory_set_shared()</code></a>. When a factory is shared,
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-construct" title="gst_rtsp_media_factory_construct ()"><code class="function">gst_rtsp_media_factory_construct()</code></a> will return the same <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> when
+the url matches.</p>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-new"></a><h3>gst_rtsp_media_factory_new ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="returnvalue">GstRTSPMediaFactory</span></a> *
+gst_rtsp_media_factory_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Create a new <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> instance.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.2.5"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> object. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-get-launch"></a><h3>gst_rtsp_media_factory_get_launch ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_media_factory_get_launch (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
+<p>Get the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstParse.html#gst-parse-launch"><code class="function">gst_parse_launch()</code></a> pipeline description that will be used in the
+default prepare vmethod.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.3.6"></a><h4>Returns</h4>
+<p> the configured launch description. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-set-launch"></a><h3>gst_rtsp_media_factory_set_launch ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_set_launch (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *launch</code></em>);</pre>
+<p>The <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstParse.html#gst-parse-launch"><code class="function">gst_parse_launch()</code></a> line to use for constructing the pipeline in the
+default prepare vmethod.</p>
+<p>The pipeline description should return a GstBin as the toplevel element
+which can be accomplished by enclosing the dscription with brackets '('
+')'.</p>
+<p>The description should return a pipeline with payloaders named pay0, pay1,
+etc.. Each of the payloaders will result in a stream.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.4.7"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>launch</p></td>
+<td class="parameter_description"><p>the launch description</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-get-permissions"></a><h3>gst_rtsp_media_factory_get_permissions ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
+gst_rtsp_media_factory_get_permissions
+ (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
+<p>Get the permissions object from <em class="parameter"><code>factory</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.5.6"></a><h4>Returns</h4>
+<p> a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> object, unref after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-set-permissions"></a><h3>gst_rtsp_media_factory_set_permissions ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_set_permissions
+ (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>);</pre>
+<p>Set <em class="parameter"><code>permissions</code></em>
+ on <em class="parameter"><code>factory</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>permissions</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-add-role"></a><h3>gst_rtsp_media_factory_add_role ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_add_role (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *fieldname</code></em>,
+ <em class="parameter"><code>...</code></em>);</pre>
+<p>A convenience method to add <em class="parameter"><code>role</code></em>
+ with <em class="parameter"><code>fieldname</code></em>
+ and additional arguments to
+the permissions of <em class="parameter"><code>factory</code></em>
+. If <em class="parameter"><code>factory</code></em>
+ had no permissions, new permissions
+will be created and the role will be added to it.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>role</p></td>
+<td class="parameter_description"><p>a role</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>fieldname</p></td>
+<td class="parameter_description"><p>the first field name</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>...</p></td>
+<td class="parameter_description"><p>additional arguments</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-set-shared"></a><h3>gst_rtsp_media_factory_set_shared ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_set_shared (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> shared</code></em>);</pre>
+<p>Configure if media created from this factory can be shared between clients.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>shared</p></td>
+<td class="parameter_description"><p>the new value</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-is-shared"></a><h3>gst_rtsp_media_factory_is_shared ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_factory_is_shared (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
+<p>Get if media created from this factory can be shared between clients.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.9.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media will be shared between clients.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-is-eos-shutdown"></a><h3>gst_rtsp_media_factory_is_eos_shutdown ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_media_factory_is_eos_shutdown
+ (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
+<p>Get if media created from this factory will have an EOS event sent to the
+pipeline before shutdown.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.10.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media will receive EOS before shutdown.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-set-eos-shutdown"></a><h3>gst_rtsp_media_factory_set_eos_shutdown ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_set_eos_shutdown
+ (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> eos_shutdown</code></em>);</pre>
+<p>Configure if media created from this factory will have an EOS sent to the
+pipeline before shutdown.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.11.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>eos_shutdown</p></td>
+<td class="parameter_description"><p>the new value</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-get-protocols"></a><h3>gst_rtsp_media_factory_get_protocols ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
+gst_rtsp_media_factory_get_protocols (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
+<p>Get the allowed protocols of <em class="parameter"><code>factory</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.12.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.12.6"></a><h4>Returns</h4>
+<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-set-protocols"></a><h3>gst_rtsp_media_factory_set_protocols ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_set_protocols (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a> protocols</code></em>);</pre>
+<p>Configure the allowed lower transport for <em class="parameter"><code>factory</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.13.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>protocols</p></td>
+<td class="parameter_description"><p>the new flags</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-set-profiles"></a><h3>gst_rtsp_media_factory_set_profiles ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_set_profiles (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a> profiles</code></em>);</pre>
+<p>Configure the allowed profiles for <em class="parameter"><code>factory</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.14.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>profiles</p></td>
+<td class="parameter_description"><p>the new flags</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-get-profiles"></a><h3>gst_rtsp_media_factory_get_profiles ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
+gst_rtsp_media_factory_get_profiles (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
+<p>Get the allowed profiles of <em class="parameter"><code>factory</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.15.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.15.6"></a><h4>Returns</h4>
+<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-get-address-pool"></a><h3>gst_rtsp_media_factory_get_address_pool ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
+gst_rtsp_media_factory_get_address_pool
+ (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
+<p>Get the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> used as the address pool of <em class="parameter"><code>factory</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.16.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.16.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> of <em class="parameter"><code>factory</code></em>
+. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-set-address-pool"></a><h3>gst_rtsp_media_factory_set_address_pool ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_set_address_pool
+ (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
+<p>configure <em class="parameter"><code>pool</code></em>
+ to be used as the address pool of <em class="parameter"><code>factory</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.17.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-get-buffer-size"></a><h3>gst_rtsp_media_factory_get_buffer_size ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_media_factory_get_buffer_size
+ (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
+<p>Get the kernel UDP buffer size.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.18.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.18.6"></a><h4>Returns</h4>
+<p> the kernel UDP buffer size.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-set-buffer-size"></a><h3>gst_rtsp_media_factory_set_buffer_size ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_set_buffer_size
+ (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> size</code></em>);</pre>
+<p>Set the kernel UDP buffer size.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.19.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>size</p></td>
+<td class="parameter_description"><p>the new value</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-get-suspend-mode"></a><h3>gst_rtsp_media_factory_get_suspend_mode ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="returnvalue">GstRTSPSuspendMode</span></a>
+gst_rtsp_media_factory_get_suspend_mode
+ (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
+<p>Get how media created from this factory will be suspended.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.20.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.20.6"></a><h4>Returns</h4>
+<p> a <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a>.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-set-suspend-mode"></a><h3>gst_rtsp_media_factory_set_suspend_mode ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_set_suspend_mode
+ (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a> mode</code></em>);</pre>
+<p>Configure how media created from this factory will be suspended.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.21.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>mode</p></td>
+<td class="parameter_description"><p>the new <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-construct"></a><h3>gst_rtsp_media_factory_construct ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
+gst_rtsp_media_factory_construct (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code>const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a> *url</code></em>);</pre>
+<p>Construct the media object and create its streams. Implementations
+should create the needed gstreamer elements and add them to the result
+object. No state changes should be performed on them yet.</p>
+<p>One or more GstRTSPStream objects should be created from the result
+with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-create-stream" title="gst_rtsp_media_create_stream ()"><code class="function">gst_rtsp_media_create_stream()</code></a>.</p>
+<p>After the media is constructed, it can be configured and then prepared
+with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.22.7"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>url</p></td>
+<td class="parameter_description"><p>the url used</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.22.8"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> if the media could be prepared. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-create-element"></a><h3>gst_rtsp_media_factory_create_element ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="returnvalue">GstElement</span></a> *
+gst_rtsp_media_factory_create_element (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
+ <em class="parameter"><code>const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a> *url</code></em>);</pre>
+<p>Construct and return a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> that is a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> containing
+the elements to use for streaming the media.</p>
+<p>The bin should contain payloaders pay%d for each stream. The default
+implementation of this function returns the bin created from the
+launch parameter.</p>
+<div class="refsect3">
+<a name="id-1.2.5.9.23.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>url</p></td>
+<td class="parameter_description"><p>the url used</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.5.9.23.7"></a><h4>Returns</h4>
+<p> a new <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a>. </p>
+<p><span class="annotation">[<acronym title="Alias for transfer none, used for objects with floating refs."><span class="acronym">transfer floating</span></acronym>]</span></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPMediaFactory-struct"></a><h3>struct GstRTSPMediaFactory</h3>
+<pre class="programlisting">struct GstRTSPMediaFactory;</pre>
+<p>The definition and logic for constructing the pipeline for a media. The media
+can contain multiple streams like audio and video.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaFactoryClass"></a><h3>struct GstRTSPMediaFactoryClass</h3>
+<pre class="programlisting">struct GstRTSPMediaFactoryClass {
+ GObjectClass parent_class;
+
+ gchar * (*gen_key) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+
+ GstElement * (*create_element) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+ GstRTSPMedia * (*construct) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+ GstElement * (*create_pipeline) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+ void (*configure) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+
+ /* signals */
+ void (*media_constructed) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+ void (*media_configure) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+};
+</pre>
+<p>The <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> class structure.</p>
+<div class="refsect3">
+<a name="id-1.2.5.10.3.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPMediaFactoryClass.parent-class"></a>parent_class</code></em>;</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.gen-key"></a>gen_key</code></em> ()</p></td>
+<td class="struct_member_description"><p>convert <em class="parameter"><code>url</code></em>
+to a key for caching shared <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> objects.
+The default implementation of this function will use the complete URL
+including the query parameters to return a key.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.create-element"></a>create_element</code></em> ()</p></td>
+<td class="struct_member_description"><p>Construct and return a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> that is a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> containing
+the elements to use for streaming the media. The bin should contain
+payloaders pay%d for each stream. The default implementation of this
+function returns the bin created from the launch parameter.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.construct"></a>construct</code></em> ()</p></td>
+<td class="struct_member_description"><p>the vmethod that will be called when the factory has to create the
+<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> for <em class="parameter"><code>url</code></em>
+. The default implementation of this
+function calls create_element to retrieve an element and then looks for
+pay%d to create the streams.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.create-pipeline"></a>create_pipeline</code></em> ()</p></td>
+<td class="struct_member_description"><p>create a new pipeline or re-use an existing one and
+add the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>'s element created by <em class="parameter"><code>construct</code></em>
+to the pipeline.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.configure"></a>configure</code></em> ()</p></td>
+<td class="struct_member_description"><p>configure the media created with <em class="parameter"><code>construct</code></em>
+. The default
+implementation will configure the 'shared' property of the media.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.media-constructed"></a>media_constructed</code></em> ()</p></td>
+<td class="struct_member_description"><p>signal emited when a media was constructed</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.media-configure"></a>media_configure</code></em> ()</p></td>
+<td class="struct_member_description"><p>signal emited when a media should be configured</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.property-details"></a><h2>Property Details</h2>
+<div class="refsect2">
+<a name="GstRTSPMediaFactory--buffer-size"></a><h3>The <code class="literal">“buffer-size”</code> property</h3>
+<pre class="programlisting"> “buffer-size” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
+<p>The kernel UDP buffer size to use.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: 524288</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaFactory--eos-shutdown"></a><h3>The <code class="literal">“eos-shutdown”</code> property</h3>
+<pre class="programlisting"> “eos-shutdown” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
+<p>Send EOS down the pipeline before shutting down.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: FALSE</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaFactory--launch"></a><h3>The <code class="literal">“launch”</code> property</h3>
+<pre class="programlisting"> “launch” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</pre>
+<p>A launch description of the pipeline.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: NULL</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaFactory--profiles"></a><h3>The <code class="literal">“profiles”</code> property</h3>
+<pre class="programlisting"> “profiles” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></pre>
+<p>Allowed transfer profiles.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: GST_RTSP_PROFILE_AVP</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaFactory--protocols"></a><h3>The <code class="literal">“protocols”</code> property</h3>
+<pre class="programlisting"> “protocols” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></pre>
+<p>Allowed lower transport protocols.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaFactory--shared"></a><h3>The <code class="literal">“shared”</code> property</h3>
+<pre class="programlisting"> “shared” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
+<p>If media from this factory is shared.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: FALSE</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaFactory--suspend-mode"></a><h3>The <code class="literal">“suspend-mode”</code> property</h3>
+<pre class="programlisting"> “suspend-mode” <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></pre>
+<p>Control how media will be suspended.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: GST_RTSP_SUSPEND_MODE_NONE</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.signal-details"></a><h2>Signal Details</h2>
+<div class="refsect2">
+<a name="GstRTSPMediaFactory-media-configure"></a><h3>The <code class="literal">“media-configure”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *gstrtspmediafactory,
+ <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaFactory-media-constructed"></a><h3>The <code class="literal">“media-constructed”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *gstrtspmediafactory,
+ <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMediaFactory.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>, <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p>
+</div>
+</div>
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+<td><a accesskey="h" href="index.html"><img src="home.png" width="16" height="16" border="0" alt="Home"></a></td>
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+<td><a accesskey="p" href="gst-rtsp-server-GstRTSPContext.html"><img src="left.png" width="16" height="16" border="0" alt="Prev"></a></td>
+<td><a accesskey="n" href="GstRTSPMediaFactory.html"><img src="right.png" width="16" height="16" border="0" alt="Next"></a></td>
+</tr></table>
+<div class="refentry">
+<a name="GstRTSPMountPoints"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="GstRTSPMountPoints.top_of_page"></a>GstRTSPMountPoints</span></h2>
+<p>GstRTSPMountPoints — Map a path to media</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="GstRTSPMountPoints.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-new" title="gst_rtsp_mount_points_new ()">gst_rtsp_mount_points_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-add-factory" title="gst_rtsp_mount_points_add_factory ()">gst_rtsp_mount_points_add_factory</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-remove-factory" title="gst_rtsp_mount_points_remove_factory ()">gst_rtsp_mount_points_remove_factory</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="returnvalue">GstRTSPMediaFactory</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-match" title="gst_rtsp_mount_points_match ()">gst_rtsp_mount_points_match</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-make-path" title="gst_rtsp_mount_points_make_path ()">gst_rtsp_mount_points_make_path</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMountPoints.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPMountPoints.html#GstRTSPMountPoints-struct" title="struct GstRTSPMountPoints">GstRTSPMountPoints</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPMountPoints.html#GstRTSPMountPointsClass" title="struct GstRTSPMountPointsClass">GstRTSPMountPointsClass</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMountPoints.object-hierarchy"></a><h2>Object Hierarchy</h2>
+<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+ <span class="lineart">╰──</span> GstRTSPMountPoints
+</pre>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMountPoints.description"></a><h2>Description</h2>
+<p>A <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> object maintains a relation between paths
+and <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> objects. This object is usually given to
+<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> and used to find the media attached to a path.</p>
+<p>With <a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-add-factory" title="gst_rtsp_mount_points_add_factory ()"><code class="function">gst_rtsp_mount_points_add_factory()</code></a> and
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-remove-factory" title="gst_rtsp_mount_points_remove_factory ()"><code class="function">gst_rtsp_mount_points_remove_factory()</code></a>, factories can be added and
+removed.</p>
+<p>With <a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-match" title="gst_rtsp_mount_points_match ()"><code class="function">gst_rtsp_mount_points_match()</code></a> you can find the <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>
+object that completely matches the given path.</p>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMountPoints.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-mount-points-new"></a><h3>gst_rtsp_mount_points_new ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
+gst_rtsp_mount_points_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Make a new mount points object.</p>
+<div class="refsect3">
+<a name="id-1.2.4.7.2.5"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-mount-points-add-factory"></a><h3>gst_rtsp_mount_points_add_factory ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_mount_points_add_factory (<em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
+<p>Attach <em class="parameter"><code>factory</code></em>
+ to the mount point <em class="parameter"><code>path</code></em>
+ in <em class="parameter"><code>mounts</code></em>
+.</p>
+<p><em class="parameter"><code>path</code></em>
+ is of the form (/node)+. Any previous mount point will be freed.</p>
+<p>Ownership is taken of the reference on <em class="parameter"><code>factory</code></em>
+ so that <em class="parameter"><code>factory</code></em>
+ should not be
+used after calling this function.</p>
+<div class="refsect3">
+<a name="id-1.2.4.7.3.7"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>mounts</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>path</p></td>
+<td class="parameter_description"><p>a mount point</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-mount-points-remove-factory"></a><h3>gst_rtsp_mount_points_remove_factory ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_mount_points_remove_factory (<em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>);</pre>
+<p>Remove the <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> associated with <em class="parameter"><code>path</code></em>
+ in <em class="parameter"><code>mounts</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.4.7.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>mounts</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>path</p></td>
+<td class="parameter_description"><p>a mount point</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-mount-points-match"></a><h3>gst_rtsp_mount_points_match ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="returnvalue">GstRTSPMediaFactory</span></a> *
+gst_rtsp_mount_points_match (<em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> *matched</code></em>);</pre>
+<p>Find the factory in <em class="parameter"><code>mounts</code></em>
+ that has the longest match with <em class="parameter"><code>path</code></em>
+.</p>
+<p>If <em class="parameter"><code>matched</code></em>
+ is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <em class="parameter"><code>path</code></em>
+ will match the factory exactly otherwise
+the amount of characters that matched is returned in <em class="parameter"><code>matched</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.4.7.5.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>mounts</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>path</p></td>
+<td class="parameter_description"><p>a mount point</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>matched</p></td>
+<td class="parameter_description"><p> the amount of <em class="parameter"><code>path</code></em>
+matched. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.4.7.5.7"></a><h4>Returns</h4>
+<p> the <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> for <em class="parameter"><code>path</code></em>
+.
+<a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-mount-points-make-path"></a><h3>gst_rtsp_mount_points_make_path ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_mount_points_make_path (<em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>,
+ <em class="parameter"><code>const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a> *url</code></em>);</pre>
+<p>Make a path string from <em class="parameter"><code>url</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.4.7.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>mounts</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>url</p></td>
+<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.4.7.6.6"></a><h4>Returns</h4>
+<p> a path string for <em class="parameter"><code>url</code></em>
+, <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMountPoints.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPMountPoints-struct"></a><h3>struct GstRTSPMountPoints</h3>
+<pre class="programlisting">struct GstRTSPMountPoints;</pre>
+<p>Creates a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> object for a given url.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMountPointsClass"></a><h3>struct GstRTSPMountPointsClass</h3>
+<pre class="programlisting">struct GstRTSPMountPointsClass {
+ GObjectClass parent_class;
+
+ gchar * (*make_path) (GstRTSPMountPoints *mounts,
+ const GstRTSPUrl *url);
+};
+</pre>
+<p>The class for the media mounts object.</p>
+<div class="refsect3">
+<a name="id-1.2.4.8.3.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPMountPointsClass.parent-class"></a>parent_class</code></em>;</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMountPointsClass.make-path"></a>make_path</code></em> ()</p></td>
+<td class="struct_member_description"><p>make a path from the given url.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPMountPoints.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>, <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
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+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#GstRTSPServer.description" class="shortcut">Description</a></span><span id="nav_hierarchy"> <span class="dim">|</span>
+ <a href="#GstRTSPServer.object-hierarchy" class="shortcut">Object Hierarchy</a></span><span id="nav_properties"> <span class="dim">|</span>
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+<a name="GstRTSPServer"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="GstRTSPServer.top_of_page"></a>GstRTSPServer</span></h2>
+<p>GstRTSPServer — The main server object</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="GstRTSPServer.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="returnvalue">GstRTSPServer</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-new" title="gst_rtsp_server_new ()">gst_rtsp_server_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-address" title="gst_rtsp_server_get_address ()">gst_rtsp_server_get_address</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-address" title="gst_rtsp_server_set_address ()">gst_rtsp_server_set_address</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-service" title="gst_rtsp_server_get_service ()">gst_rtsp_server_get_service</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-service" title="gst_rtsp_server_set_service ()">gst_rtsp_server_set_service</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-backlog" title="gst_rtsp_server_get_backlog ()">gst_rtsp_server_get_backlog</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-backlog" title="gst_rtsp_server_set_backlog ()">gst_rtsp_server_set_backlog</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">int</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-bound-port" title="gst_rtsp_server_get_bound_port ()">gst_rtsp_server_get_bound_port</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-mount-points" title="gst_rtsp_server_get_mount_points ()">gst_rtsp_server_get_mount_points</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-mount-points" title="gst_rtsp_server_set_mount_points ()">gst_rtsp_server_set_mount_points</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-session-pool" title="gst_rtsp_server_get_session_pool ()">gst_rtsp_server_get_session_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-session-pool" title="gst_rtsp_server_set_session_pool ()">gst_rtsp_server_set_session_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-thread-pool" title="gst_rtsp_server_get_thread_pool ()">gst_rtsp_server_get_thread_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-thread-pool" title="gst_rtsp_server_set_thread_pool ()">gst_rtsp_server_set_thread_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-auth" title="gst_rtsp_server_get_auth ()">gst_rtsp_server_get_auth</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-auth" title="gst_rtsp_server_set_auth ()">gst_rtsp_server_set_auth</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-transfer-connection" title="gst_rtsp_server_transfer_connection ()">gst_rtsp_server_transfer_connection</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-io-func" title="gst_rtsp_server_io_func ()">gst_rtsp_server_io_func</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-socket" title="gst_rtsp_server_create_socket ()">gst_rtsp_server_create_socket</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="returnvalue">GSource</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-source" title="gst_rtsp_server_create_source ()">gst_rtsp_server_create_source</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-attach" title="gst_rtsp_server_attach ()">gst_rtsp_server_attach</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
+</td>
+<td class="function_name">
+<span class="c_punctuation">(</span><a class="link" href="GstRTSPServer.html#GstRTSPServerClientFilterFunc" title="GstRTSPServerClientFilterFunc ()">*GstRTSPServerClientFilterFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-client-filter" title="gst_rtsp_server_client_filter ()">gst_rtsp_server_client_filter</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPServer.properties"></a><h2>Properties</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="properties_type">
+<col width="300px" class="properties_name">
+<col width="200px" class="properties_flags">
+</colgroup>
+<tbody>
+<tr>
+<td class="property_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</td>
+<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--address" title="The “address” property">address</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--backlog" title="The “backlog” property">backlog</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--bound-port" title="The “bound-port” property">bound-port</a></td>
+<td class="property_flags">Read</td>
+</tr>
+<tr>
+<td class="property_type">
+<a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *</td>
+<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--mount-points" title="The “mount-points” property">mount-points</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</td>
+<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--service" title="The “service” property">service</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type">
+<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *</td>
+<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--session-pool" title="The “session-pool” property">session-pool</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPServer.signals"></a><h2>Signals</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="signals_return">
+<col width="300px" class="signals_name">
+<col width="200px" class="signals_flags">
+</colgroup>
+<tbody><tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer-client-connected" title="The “client-connected” signal">client-connected</a></td>
+<td class="signal_flags">Run Last</td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPServer.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer-struct" title="struct GstRTSPServer">GstRTSPServer</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPServer.html#GstRTSPServerClass" title="struct GstRTSPServerClass">GstRTSPServerClass</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPServer.object-hierarchy"></a><h2>Object Hierarchy</h2>
+<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+ <span class="lineart">╰──</span> GstRTSPServer
+</pre>
+</div>
+<div class="refsect1">
+<a name="GstRTSPServer.description"></a><h2>Description</h2>
+<p>The server object is the object listening for connections on a port and
+creating <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> objects to handle those connections.</p>
+<p>The server will listen on the address set with <a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-address" title="gst_rtsp_server_set_address ()"><code class="function">gst_rtsp_server_set_address()</code></a>
+and the port or service configured with <a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-service" title="gst_rtsp_server_set_service ()"><code class="function">gst_rtsp_server_set_service()</code></a>.
+Use <a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-backlog" title="gst_rtsp_server_set_backlog ()"><code class="function">gst_rtsp_server_set_backlog()</code></a> to configure the amount of pending requests
+that the server will keep. By default the server listens on the current
+network (0.0.0.0) and port 8554.</p>
+<p>The server will require an SSL connection when a TLS certificate has been
+set in the auth object with <a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate" title="gst_rtsp_auth_set_tls_certificate ()"><code class="function">gst_rtsp_auth_set_tls_certificate()</code></a>.</p>
+<p>To start the server, use <a class="link" href="GstRTSPServer.html#gst-rtsp-server-attach" title="gst_rtsp_server_attach ()"><code class="function">gst_rtsp_server_attach()</code></a> to attach it to a
+<a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a>. For more control, <a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-source" title="gst_rtsp_server_create_source ()"><code class="function">gst_rtsp_server_create_source()</code></a> and
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-socket" title="gst_rtsp_server_create_socket ()"><code class="function">gst_rtsp_server_create_socket()</code></a> can be used to get a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="type">GSource</span></a> and <a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a>
+respectively.</p>
+<p>gst_rtsp_server_transfer_connection() can be used to transfer an existing
+socket to the RTSP server, for example from an HTTP server.</p>
+<p>Once the server socket is attached to a mainloop, it will start accepting
+connections. When a new connection is received, a new <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> object
+is created to handle the connection. The new client will be configured with
+the server <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>, <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>, <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> and
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a>.</p>
+<p>The server uses the configured <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> object to handle the
+remainder of the communication with this client.</p>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="GstRTSPServer.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-server-new"></a><h3>gst_rtsp_server_new ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="returnvalue">GstRTSPServer</span></a> *
+gst_rtsp_server_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Create a new <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> instance.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.2.5"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-get-address"></a><h3>gst_rtsp_server_get_address ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_server_get_address (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
+<p>Get the address on which the server will accept connections.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.3.6"></a><h4>Returns</h4>
+<p> the server address. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-set-address"></a><h3>gst_rtsp_server_set_address ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_server_set_address (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *address</code></em>);</pre>
+<p>Configure <em class="parameter"><code>server</code></em>
+ to accept connections on the given address.</p>
+<p>This function must be called before the server is bound.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.4.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>address</p></td>
+<td class="parameter_description"><p>the address</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-get-service"></a><h3>gst_rtsp_server_get_service ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_server_get_service (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
+<p>Get the service on which the server will accept connections.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.5.6"></a><h4>Returns</h4>
+<p> the service. use <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-set-service"></a><h3>gst_rtsp_server_set_service ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_server_set_service (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *service</code></em>);</pre>
+<p>Configure <em class="parameter"><code>server</code></em>
+ to accept connections on the given service.
+<em class="parameter"><code>service</code></em>
+ should be a string containing the service name (see services(5)) or
+a string containing a port number between 1 and 65535.</p>
+<p>When <em class="parameter"><code>service</code></em>
+ is set to "0", the server will listen on a random free
+port. The actual used port can be retrieved with
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-bound-port" title="gst_rtsp_server_get_bound_port ()"><code class="function">gst_rtsp_server_get_bound_port()</code></a>.</p>
+<p>This function must be called before the server is bound.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.6.7"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>service</p></td>
+<td class="parameter_description"><p>the service</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-get-backlog"></a><h3>gst_rtsp_server_get_backlog ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
+gst_rtsp_server_get_backlog (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
+<p>The maximum amount of queued requests for the server.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.7.6"></a><h4>Returns</h4>
+<p> the server backlog.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-set-backlog"></a><h3>gst_rtsp_server_set_backlog ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_server_set_backlog (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> backlog</code></em>);</pre>
+<p>configure the maximum amount of requests that may be queued for the
+server.</p>
+<p>This function must be called before the server is bound.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.8.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>backlog</p></td>
+<td class="parameter_description"><p>the backlog</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-get-bound-port"></a><h3>gst_rtsp_server_get_bound_port ()</h3>
+<pre class="programlisting"><span class="returnvalue">int</span>
+gst_rtsp_server_get_bound_port (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
+<p>Get the port number where the server was bound to.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.9.6"></a><h4>Returns</h4>
+<p> the port number</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-get-mount-points"></a><h3>gst_rtsp_server_get_mount_points ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
+gst_rtsp_server_get_mount_points (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
+<p>Get the <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> used as the mount points of <em class="parameter"><code>server</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.10.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> of <em class="parameter"><code>server</code></em>
+. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-set-mount-points"></a><h3>gst_rtsp_server_set_mount_points ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_server_set_mount_points (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>);</pre>
+<p>configure <em class="parameter"><code>mounts</code></em>
+ to be used as the mount points of <em class="parameter"><code>server</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.11.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>mounts</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-get-session-pool"></a><h3>gst_rtsp_server_get_session_pool ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
+gst_rtsp_server_get_session_pool (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
+<p>Get the <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> used as the session pool of <em class="parameter"><code>server</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.12.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.12.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> used for sessions. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-set-session-pool"></a><h3>gst_rtsp_server_set_session_pool ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_server_set_session_pool (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
+<p>configure <em class="parameter"><code>pool</code></em>
+ to be used as the session pool of <em class="parameter"><code>server</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.13.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-get-thread-pool"></a><h3>gst_rtsp_server_get_thread_pool ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
+gst_rtsp_server_get_thread_pool (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
+<p>Get the <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> used as the thread pool of <em class="parameter"><code>server</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.14.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.14.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> of <em class="parameter"><code>server</code></em>
+. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-set-thread-pool"></a><h3>gst_rtsp_server_set_thread_pool ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_server_set_thread_pool (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> *pool</code></em>);</pre>
+<p>configure <em class="parameter"><code>pool</code></em>
+ to be used as the thread pool of <em class="parameter"><code>server</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.15.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-get-auth"></a><h3>gst_rtsp_server_get_auth ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
+gst_rtsp_server_get_auth (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
+<p>Get the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> used as the authentication manager of <em class="parameter"><code>server</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.16.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.16.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> of <em class="parameter"><code>server</code></em>
+. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-set-auth"></a><h3>gst_rtsp_server_set_auth ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_server_set_auth (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>);</pre>
+<p>configure <em class="parameter"><code>auth</code></em>
+ to be used as the authentication manager of <em class="parameter"><code>server</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.17.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>auth</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-transfer-connection"></a><h3>gst_rtsp_server_transfer_connection ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_server_transfer_connection (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a> *socket</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *ip</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> port</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *initial_buffer</code></em>);</pre>
+<p>Take an existing network socket and use it for an RTSP connection. This
+is used when transferring a socket from an HTTP server which should be used
+as an RTSP over HTTP tunnel. The <em class="parameter"><code>initial_buffer</code></em>
+ contains any remaining data
+that the HTTP server read from the socket while parsing the HTTP header.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.18.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>socket</p></td>
+<td class="parameter_description"><p> a network socket. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>ip</p></td>
+<td class="parameter_description"><p>the IP address of the remote client</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>port</p></td>
+<td class="parameter_description"><p>the port used by the other end</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>initial_buffer</p></td>
+<td class="parameter_description"><p>any initial data that was already read from the socket</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.18.6"></a><h4>Returns</h4>
+<p> TRUE if all was ok, FALSE if an error occurred.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-io-func"></a><h3>gst_rtsp_server_io_func ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_server_io_func (<em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a> *socket</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-IO-Channels.html#GIOCondition"><span class="type">GIOCondition</span></a> condition</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
+<p>A default <a href="https://developer.gnome.org/gio/unstable/GSocket.html#GSocketSourceFunc"><span class="type">GSocketSourceFunc</span></a> that creates a new <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> to accept and handle a
+new connection on <em class="parameter"><code>socket</code></em>
+ or <em class="parameter"><code>server</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.19.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>socket</p></td>
+<td class="parameter_description"><p>a <a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>condition</p></td>
+<td class="parameter_description"><p>the condition on <em class="parameter"><code>source</code></em>
+</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.19.6"></a><h4>Returns</h4>
+<p> TRUE if the source could be connected, FALSE if an error occurred.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-create-socket"></a><h3>gst_rtsp_server_create_socket ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
+gst_rtsp_server_create_socket (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GCancellable.html"><span class="type">GCancellable</span></a> *cancellable</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Error-Reporting.html#GError"><span class="type">GError</span></a> **error</code></em>);</pre>
+<p>Create a <a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a> for <em class="parameter"><code>server</code></em>
+. The socket will listen on the
+configured service.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.20.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>cancellable</p></td>
+<td class="parameter_description"><p> a <a href="https://developer.gnome.org/gio/unstable/GCancellable.html"><span class="type">GCancellable</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>error</p></td>
+<td class="parameter_description"><p> a <a href="https://developer.gnome.org/glib/unstable/glib-Error-Reporting.html#GError"><span class="type">GError</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.20.6"></a><h4>Returns</h4>
+<p> the <a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a> for <em class="parameter"><code>server</code></em>
+or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when an error
+occurred. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-create-source"></a><h3>gst_rtsp_server_create_source ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="returnvalue">GSource</span></a> *
+gst_rtsp_server_create_source (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GCancellable.html"><span class="type">GCancellable</span></a> *cancellable</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Error-Reporting.html#GError"><span class="type">GError</span></a> **error</code></em>);</pre>
+<p>Create a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="type">GSource</span></a> for <em class="parameter"><code>server</code></em>
+. The new source will have a default
+<a href="https://developer.gnome.org/gio/unstable/GSocket.html#GSocketSourceFunc"><span class="type">GSocketSourceFunc</span></a> of <a class="link" href="GstRTSPServer.html#gst-rtsp-server-io-func" title="gst_rtsp_server_io_func ()"><code class="function">gst_rtsp_server_io_func()</code></a>.</p>
+<p><em class="parameter"><code>cancellable</code></em>
+ if not <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> can be used to cancel the source, which will cause
+the source to trigger, reporting the current condition (which is likely 0
+unless cancellation happened at the same time as a condition change). You can
+check for this in the callback using <a href="https://developer.gnome.org/gio/unstable/GCancellable.html#g-cancellable-is-cancelled"><code class="function">g_cancellable_is_cancelled()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.21.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>cancellable</p></td>
+<td class="parameter_description"><p> a <a href="https://developer.gnome.org/gio/unstable/GCancellable.html"><span class="type">GCancellable</span></a> or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>error</p></td>
+<td class="parameter_description"><p> a <a href="https://developer.gnome.org/glib/unstable/glib-Error-Reporting.html#GError"><span class="type">GError</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.21.7"></a><h4>Returns</h4>
+<p> the <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="type">GSource</span></a> for <em class="parameter"><code>server</code></em>
+or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when an error
+occurred. Free with <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#g-source-unref"><code class="function">g_source_unref()</code></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-attach"></a><h3>gst_rtsp_server_attach ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_server_attach (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a> *context</code></em>);</pre>
+<p>Attaches <em class="parameter"><code>server</code></em>
+ to <em class="parameter"><code>context</code></em>
+. When the mainloop for <em class="parameter"><code>context</code></em>
+ is run, the
+server will be dispatched. When <em class="parameter"><code>context</code></em>
+ is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, the default context will be
+used).</p>
+<p>This function should be called when the server properties and urls are fully
+configured and the server is ready to start.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.22.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>context</p></td>
+<td class="parameter_description"><p> a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.22.7"></a><h4>Returns</h4>
+<p> the ID (greater than 0) for the source within the GMainContext.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPServerClientFilterFunc"></a><h3>GstRTSPServerClientFilterFunc ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
+<span class="c_punctuation">(</span>*GstRTSPServerClientFilterFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>This function will be called by the <a class="link" href="GstRTSPServer.html#gst-rtsp-server-client-filter" title="gst_rtsp_server_client_filter ()"><code class="function">gst_rtsp_server_client_filter()</code></a>. An
+implementation should return a value of <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
+<p>When this function returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, <em class="parameter"><code>client</code></em>
+ will be removed
+from <em class="parameter"><code>server</code></em>
+.</p>
+<p>A return value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a> will leave <em class="parameter"><code>client</code></em>
+ untouched in
+<em class="parameter"><code>server</code></em>
+.</p>
+<p>A value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will add <em class="parameter"><code>client</code></em>
+ to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-client-filter" title="gst_rtsp_server_client_filter ()"><code class="function">gst_rtsp_server_client_filter()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.23.8"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> object</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> in <em class="parameter"><code>server</code></em>
+</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data that has been given to <a class="link" href="GstRTSPServer.html#gst-rtsp-server-client-filter" title="gst_rtsp_server_client_filter ()"><code class="function">gst_rtsp_server_client_filter()</code></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.23.9"></a><h4>Returns</h4>
+<p> a <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-server-client-filter"></a><h3>gst_rtsp_server_client_filter ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
+gst_rtsp_server_client_filter (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPServer.html#GstRTSPServerClientFilterFunc" title="GstRTSPServerClientFilterFunc ()"><span class="type">GstRTSPServerClientFilterFunc</span></a> func</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>Call <em class="parameter"><code>func</code></em>
+ for each client managed by <em class="parameter"><code>server</code></em>
+. The result value of <em class="parameter"><code>func</code></em>
+
+determines what happens to the client. <em class="parameter"><code>func</code></em>
+ will be called with <em class="parameter"><code>server</code></em>
+
+locked so no further actions on <em class="parameter"><code>server</code></em>
+ can be performed from <em class="parameter"><code>func</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, the client will be removed from
+<em class="parameter"><code>server</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a>, the client will remain in <em class="parameter"><code>server</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>, the client will remain in <em class="parameter"><code>server</code></em>
+ but
+will also be added with an additional ref to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of this
+function..</p>
+<p>When <em class="parameter"><code>func</code></em>
+ is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will be assumed for each client.</p>
+<div class="refsect3">
+<a name="id-1.2.1.9.24.9"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>server</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>func</p></td>
+<td class="parameter_description"><p> a callback. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid only during the call to the method."><span class="acronym">scope call</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data passed to <em class="parameter"><code>func</code></em>
+</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.1.9.24.10"></a><h4>Returns</h4>
+<p> a <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> with all
+clients for which <em class="parameter"><code>func</code></em>
+returned <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>. After usage, each
+element in the <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> should be unreffed before the list is freed. </p>
+<p><span class="annotation">[<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRTSPClient][<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPServer.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPServer-struct"></a><h3>struct GstRTSPServer</h3>
+<pre class="programlisting">struct GstRTSPServer;</pre>
+<p>This object listens on a port, creates and manages the clients connected to
+it.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPServerClass"></a><h3>struct GstRTSPServerClass</h3>
+<pre class="programlisting">struct GstRTSPServerClass {
+ GObjectClass parent_class;
+
+ GstRTSPClient * (*create_client) (GstRTSPServer *server);
+
+ /* signals */
+ void (*client_connected) (GstRTSPServer *server, GstRTSPClient *client);
+};
+</pre>
+<p>The RTSP server class structure</p>
+<div class="refsect3">
+<a name="id-1.2.1.10.3.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPServerClass.parent-class"></a>parent_class</code></em>;</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPServerClass.create-client"></a>create_client</code></em> ()</p></td>
+<td class="struct_member_description"><p>Create, configure a new GstRTSPClient
+object that handles the new connection on <em class="parameter"><code>socket</code></em>
+. The default
+implementation will create a GstRTSPClient and will configure the
+mount-points, auth, session-pool and thread-pool on the client.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPServerClass.client-connected"></a>client_connected</code></em> ()</p></td>
+<td class="struct_member_description"><p>emited when a new client connected.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPServer.property-details"></a><h2>Property Details</h2>
+<div class="refsect2">
+<a name="GstRTSPServer--address"></a><h3>The <code class="literal">“address”</code> property</h3>
+<pre class="programlisting"> “address” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</pre>
+<p>The address the server uses to listen on.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: "0.0.0.0"</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPServer--backlog"></a><h3>The <code class="literal">“backlog”</code> property</h3>
+<pre class="programlisting"> “backlog” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a></pre>
+<p>The maximum length to which the queue of pending connections may grow.</p>
+<p>Flags: Read / Write</p>
+<p>Allowed values: >= 0</p>
+<p>Default value: 5</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPServer--bound-port"></a><h3>The <code class="literal">“bound-port”</code> property</h3>
+<pre class="programlisting"> “bound-port” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a></pre>
+<p>The port number the server is listening on.</p>
+<p>Flags: Read</p>
+<p>Allowed values: [-1,65535]</p>
+<p>Default value: -1</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPServer--mount-points"></a><h3>The <code class="literal">“mount-points”</code> property</h3>
+<pre class="programlisting"> “mount-points” <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *</pre>
+<p>The mount points to use for client session.</p>
+<p>Flags: Read / Write</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPServer--service"></a><h3>The <code class="literal">“service”</code> property</h3>
+<pre class="programlisting"> “service” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</pre>
+<p>The service or port number the server uses to listen on.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: "8554"</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPServer--session-pool"></a><h3>The <code class="literal">“session-pool”</code> property</h3>
+<pre class="programlisting"> “session-pool” <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *</pre>
+<p>The session pool to use for client session.</p>
+<p>Flags: Read / Write</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPServer.signal-details"></a><h2>Signal Details</h2>
+<div class="refsect2">
+<a name="GstRTSPServer-client-connected"></a><h3>The <code class="literal">“client-connected”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *gstrtspserver,
+ <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPServer.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
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+<html>
+<head>
+<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
+<title>GStreamer RTSP Server Reference Manual: GstRTSPSession</title>
+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
+<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
+<link rel="up" href="ch01.html" title="">
+<link rel="prev" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">
+<link rel="next" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">
+<meta name="generator" content="GTK-Doc V1.21 (XML mode)">
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+<body bgcolor="white" text="black" link="#0000FF" vlink="#840084" alink="#0000FF">
+<table class="navigation" id="top" width="100%" summary="Navigation header" cellpadding="2" cellspacing="5"><tr valign="middle">
+<td width="100%" align="left" class="shortcuts">
+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#GstRTSPSession.description" class="shortcut">Description</a></span><span id="nav_hierarchy"> <span class="dim">|</span>
+ <a href="#GstRTSPSession.object-hierarchy" class="shortcut">Object Hierarchy</a></span><span id="nav_properties"> <span class="dim">|</span>
+ <a href="#GstRTSPSession.properties" class="shortcut">Properties</a></span>
+</td>
+<td><a accesskey="h" href="index.html"><img src="home.png" width="16" height="16" border="0" alt="Home"></a></td>
+<td><a accesskey="u" href="ch01.html"><img src="up.png" width="16" height="16" border="0" alt="Up"></a></td>
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+</tr></table>
+<div class="refentry">
+<a name="GstRTSPSession"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="GstRTSPSession.top_of_page"></a>GstRTSPSession</span></h2>
+<p>GstRTSPSession — An object to manage media</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="GstRTSPSession.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-new" title="gst_rtsp_session_new ()">gst_rtsp_session_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-sessionid" title="gst_rtsp_session_get_sessionid ()">gst_rtsp_session_get_sessionid</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-header" title="gst_rtsp_session_get_header ()">gst_rtsp_session_get_header</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-set-timeout" title="gst_rtsp_session_set_timeout ()">gst_rtsp_session_set_timeout</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-timeout" title="gst_rtsp_session_get_timeout ()">gst_rtsp_session_get_timeout</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-touch" title="gst_rtsp_session_touch ()">gst_rtsp_session_touch</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-prevent-expire" title="gst_rtsp_session_prevent_expire ()">gst_rtsp_session_prevent_expire</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-allow-expire" title="gst_rtsp_session_allow_expire ()">gst_rtsp_session_allow_expire</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-next-timeout" title="gst_rtsp_session_next_timeout ()">gst_rtsp_session_next_timeout</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-is-expired" title="gst_rtsp_session_is_expired ()">gst_rtsp_session_is_expired</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-manage-media" title="gst_rtsp_session_manage_media ()">gst_rtsp_session_manage_media</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-release-media" title="gst_rtsp_session_release_media ()">gst_rtsp_session_release_media</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-media" title="gst_rtsp_session_get_media ()">gst_rtsp_session_get_media</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
+</td>
+<td class="function_name">
+<span class="c_punctuation">(</span><a class="link" href="GstRTSPSession.html#GstRTSPSessionFilterFunc" title="GstRTSPSessionFilterFunc ()">*GstRTSPSessionFilterFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()">gst_rtsp_session_filter</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSession.properties"></a><h2>Properties</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="properties_type">
+<col width="300px" class="properties_name">
+<col width="200px" class="properties_flags">
+</colgroup>
+<tbody>
+<tr>
+<td class="property_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</td>
+<td class="property_name"><a class="link" href="GstRTSPSession.html#GstRTSPSession--sessionid" title="The “sessionid” property">sessionid</a></td>
+<td class="property_flags">Read / Write / Construct Only</td>
+</tr>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPSession.html#GstRTSPSession--timeout" title="The “timeout” property">timeout</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+<tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPSession.html#GstRTSPSession--timeout-always-visible" title="The “timeout-always-visible” property">timeout-always-visible</a></td>
+<td class="property_flags">Read / Write</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSession.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPSession.html#GstRTSPSession-struct" title="struct GstRTSPSession">GstRTSPSession</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPSession.html#GstRTSPSessionClass" title="struct GstRTSPSessionClass">GstRTSPSessionClass</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">enum</td>
+<td class="function_name"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult">GstRTSPFilterResult</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSession.object-hierarchy"></a><h2>Object Hierarchy</h2>
+<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+ <span class="lineart">╰──</span> GstRTSPSession
+</pre>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSession.description"></a><h2>Description</h2>
+<p>The <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> is identified by an id, unique in the
+<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> that created the session and manages media and its
+configuration.</p>
+<p>A <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> has a timeout that can be retrieved with
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-timeout" title="gst_rtsp_session_get_timeout ()"><code class="function">gst_rtsp_session_get_timeout()</code></a>. You can check if the sessions is expired with
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-is-expired" title="gst_rtsp_session_is_expired ()"><code class="function">gst_rtsp_session_is_expired()</code></a>. <a class="link" href="GstRTSPSession.html#gst-rtsp-session-touch" title="gst_rtsp_session_touch ()"><code class="function">gst_rtsp_session_touch()</code></a> will reset the
+expiration counter of the session.</p>
+<p>When a client configures a media with SETUP, a session will be created to
+keep track of the configuration of that media. With
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-manage-media" title="gst_rtsp_session_manage_media ()"><code class="function">gst_rtsp_session_manage_media()</code></a>, the media is added to the managed media
+in the session. With <a class="link" href="GstRTSPSession.html#gst-rtsp-session-release-media" title="gst_rtsp_session_release_media ()"><code class="function">gst_rtsp_session_release_media()</code></a> the media can be
+released again from the session. Managed media is identified in the sessions
+with a url. Use <a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-media" title="gst_rtsp_session_get_media ()"><code class="function">gst_rtsp_session_get_media()</code></a> to get the media that matches
+(part of) the given url.</p>
+<p>The media in a session can be iterated with <a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()"><code class="function">gst_rtsp_session_filter()</code></a>.</p>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSession.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-session-new"></a><h3>gst_rtsp_session_new ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
+gst_rtsp_session_new (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *sessionid</code></em>);</pre>
+<p>Create a new <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> instance with <em class="parameter"><code>sessionid</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.2.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>sessionid</p></td>
+<td class="parameter_description"><p>a session id</p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.2.6"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-get-sessionid"></a><h3>gst_rtsp_session_get_sessionid ()</h3>
+<pre class="programlisting">const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_session_get_sessionid (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
+<p>Get the sessionid of <em class="parameter"><code>session</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.3.6"></a><h4>Returns</h4>
+<p> the sessionid of <em class="parameter"><code>session</code></em>
+. The value remains valid
+as long as <em class="parameter"><code>session</code></em>
+is alive. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-get-header"></a><h3>gst_rtsp_session_get_header ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_session_get_header (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
+<p>Get the string that can be placed in the Session header field.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.4.6"></a><h4>Returns</h4>
+<p> the Session header of <em class="parameter"><code>session</code></em>
+. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-set-timeout"></a><h3>gst_rtsp_session_set_timeout ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_session_set_timeout (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> timeout</code></em>);</pre>
+<p>Configure <em class="parameter"><code>session</code></em>
+ for a timeout of <em class="parameter"><code>timeout</code></em>
+ seconds. The session will be
+cleaned up when there is no activity for <em class="parameter"><code>timeout</code></em>
+ seconds.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>timeout</p></td>
+<td class="parameter_description"><p>the new timeout</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-get-timeout"></a><h3>gst_rtsp_session_get_timeout ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_session_get_timeout (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
+<p>Get the timeout value of <em class="parameter"><code>session</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.6.6"></a><h4>Returns</h4>
+<p> the timeout of <em class="parameter"><code>session</code></em>
+in seconds.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-touch"></a><h3>gst_rtsp_session_touch ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_session_touch (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
+<p>Update the last_access time of the session to the current time.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-prevent-expire"></a><h3>gst_rtsp_session_prevent_expire ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_session_prevent_expire (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
+<p>Prevent <em class="parameter"><code>session</code></em>
+ from expiring.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-allow-expire"></a><h3>gst_rtsp_session_allow_expire ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_session_allow_expire (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
+<p>Allow <em class="parameter"><code>session</code></em>
+ to expire. This method must be called an equal
+amount of time as <a class="link" href="GstRTSPSession.html#gst-rtsp-session-prevent-expire" title="gst_rtsp_session_prevent_expire ()"><code class="function">gst_rtsp_session_prevent_expire()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-next-timeout"></a><h3>gst_rtsp_session_next_timeout ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
+gst_rtsp_session_next_timeout (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Date-and-Time-Functions.html#GTimeVal"><span class="type">GTimeVal</span></a> *now</code></em>);</pre>
+<p>Get the amount of milliseconds till the session will expire.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>now</p></td>
+<td class="parameter_description"><p> the current system time. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.10.6"></a><h4>Returns</h4>
+<p> the amount of milliseconds since the session will time out.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-is-expired"></a><h3>gst_rtsp_session_is_expired ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_session_is_expired (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Date-and-Time-Functions.html#GTimeVal"><span class="type">GTimeVal</span></a> *now</code></em>);</pre>
+<p>Check if <em class="parameter"><code>session</code></em>
+ timeout out.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.11.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>now</p></td>
+<td class="parameter_description"><p> the current system time. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.11.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>session</code></em>
+timed out</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-manage-media"></a><h3>gst_rtsp_session_manage_media ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
+gst_rtsp_session_manage_media (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Manage the media object <em class="parameter"><code>obj</code></em>
+ in <em class="parameter"><code>sess</code></em>
+. <em class="parameter"><code>path</code></em>
+ will be used to retrieve this
+media from the session with <a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-media" title="gst_rtsp_session_get_media ()"><code class="function">gst_rtsp_session_get_media()</code></a>.</p>
+<p>Ownership is taken from <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.12.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>sess</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>path</p></td>
+<td class="parameter_description"><p>the path for the media</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.12.7"></a><h4>Returns</h4>
+<p> a new <em class="parameter"><code>GstRTSPSessionMedia</code></em>
+object. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-release-media"></a><h3>gst_rtsp_session_release_media ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_session_release_media (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>);</pre>
+<p>Release the managed <em class="parameter"><code>media</code></em>
+ in <em class="parameter"><code>sess</code></em>
+, freeing the memory allocated by it.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.13.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>sess</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.13.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if there are more media session left in <em class="parameter"><code>sess</code></em>
+.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-get-media"></a><h3>gst_rtsp_session_get_media ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
+gst_rtsp_session_get_media (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> *matched</code></em>);</pre>
+<p>Get the session media for <em class="parameter"><code>path</code></em>
+. <em class="parameter"><code>matched</code></em>
+ will contain the number of matched
+characters of <em class="parameter"><code>path</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.14.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>sess</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>path</p></td>
+<td class="parameter_description"><p>the path for the media</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>matched</p></td>
+<td class="parameter_description"><p> the amount of matched characters. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.14.6"></a><h4>Returns</h4>
+<p> the configuration for <em class="parameter"><code>path</code></em>
+in <em class="parameter"><code>sess</code></em>
+. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPSessionFilterFunc"></a><h3>GstRTSPSessionFilterFunc ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
+<span class="c_punctuation">(</span>*GstRTSPSessionFilterFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>This function will be called by the <a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()"><code class="function">gst_rtsp_session_filter()</code></a>. An
+implementation should return a value of <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
+<p>When this function returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, <em class="parameter"><code>media</code></em>
+ will be removed
+from <em class="parameter"><code>sess</code></em>
+.</p>
+<p>A return value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a> will leave <em class="parameter"><code>media</code></em>
+ untouched in
+<em class="parameter"><code>sess</code></em>
+.</p>
+<p>A value of GST_RTSP_FILTER_REF will add <em class="parameter"><code>media</code></em>
+ to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()"><code class="function">gst_rtsp_session_filter()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.15.8"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>sess</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> object</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> in <em class="parameter"><code>sess</code></em>
+</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data that has been given to <a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()"><code class="function">gst_rtsp_session_filter()</code></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.15.9"></a><h4>Returns</h4>
+<p> a <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-filter"></a><h3>gst_rtsp_session_filter ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
+gst_rtsp_session_filter (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPSession.html#GstRTSPSessionFilterFunc" title="GstRTSPSessionFilterFunc ()"><span class="type">GstRTSPSessionFilterFunc</span></a> func</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>Call <em class="parameter"><code>func</code></em>
+ for each media in <em class="parameter"><code>sess</code></em>
+. The result value of <em class="parameter"><code>func</code></em>
+ determines
+what happens to the media. <em class="parameter"><code>func</code></em>
+ will be called with <em class="parameter"><code>sess</code></em>
+
+locked so no further actions on <em class="parameter"><code>sess</code></em>
+ can be performed from <em class="parameter"><code>func</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, the media will be removed from
+<em class="parameter"><code>sess</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a>, the media will remain in <em class="parameter"><code>sess</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>, the media will remain in <em class="parameter"><code>sess</code></em>
+ but
+will also be added with an additional ref to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of this
+function..</p>
+<p>When <em class="parameter"><code>func</code></em>
+ is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will be assumed for all media.</p>
+<div class="refsect3">
+<a name="id-1.2.10.8.16.9"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>sess</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>func</p></td>
+<td class="parameter_description"><p> a callback. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid only during the call to the method."><span class="acronym">scope call</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p> user data passed to <em class="parameter"><code>func</code></em>
+. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.10.8.16.10"></a><h4>Returns</h4>
+<p> a GList with all
+media for which <em class="parameter"><code>func</code></em>
+returned <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>. After usage, each
+element in the <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> should be unreffed before the list is freed. </p>
+<p><span class="annotation">[<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRTSPSessionMedia][<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSession.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPSession-struct"></a><h3>struct GstRTSPSession</h3>
+<pre class="programlisting">struct GstRTSPSession;</pre>
+<p>Session information kept by the server for a specific client.
+One client session, identified with a session id, can handle multiple medias
+identified with the url of a media.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPSessionClass"></a><h3>struct GstRTSPSessionClass</h3>
+<pre class="programlisting">struct GstRTSPSessionClass {
+ GObjectClass parent_class;
+};
+</pre>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPFilterResult"></a><h3>enum GstRTSPFilterResult</h3>
+<p>Possible return values for <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()"><code class="function">gst_rtsp_session_pool_filter()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.10.9.4.4"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="enum_members_name">
+<col class="enum_members_description">
+<col width="200px" class="enum_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-FILTER-REMOVE:CAPS"></a>GST_RTSP_FILTER_REMOVE</p></td>
+<td class="enum_member_description">
+<p>Remove session</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-FILTER-KEEP:CAPS"></a>GST_RTSP_FILTER_KEEP</p></td>
+<td class="enum_member_description">
+<p>Keep session in the pool</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-FILTER-REF:CAPS"></a>GST_RTSP_FILTER_REF</p></td>
+<td class="enum_member_description">
+<p>Ref session in the result list</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSession.property-details"></a><h2>Property Details</h2>
+<div class="refsect2">
+<a name="GstRTSPSession--sessionid"></a><h3>The <code class="literal">“sessionid”</code> property</h3>
+<pre class="programlisting"> “sessionid” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</pre>
+<p>the session id.</p>
+<p>Flags: Read / Write / Construct Only</p>
+<p>Default value: NULL</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPSession--timeout"></a><h3>The <code class="literal">“timeout”</code> property</h3>
+<pre class="programlisting"> “timeout” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
+<p>the timeout of the session (0 = never).</p>
+<p>Flags: Read / Write</p>
+<p>Default value: 60</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPSession--timeout-always-visible"></a><h3>The <code class="literal">“timeout-always-visible”</code> property</h3>
+<pre class="programlisting"> “timeout-always-visible” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
+<p>timeout always visible in header.</p>
+<p>Flags: Read / Write</p>
+<p>Default value: FALSE</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSession.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a>, <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
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+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#GstRTSPSessionPool.description" class="shortcut">Description</a></span><span id="nav_hierarchy"> <span class="dim">|</span>
+ <a href="#GstRTSPSessionPool.object-hierarchy" class="shortcut">Object Hierarchy</a></span><span id="nav_properties"> <span class="dim">|</span>
+ <a href="#GstRTSPSessionPool.properties" class="shortcut">Properties</a></span><span id="nav_signals"> <span class="dim">|</span>
+ <a href="#GstRTSPSessionPool.signals" class="shortcut">Signals</a></span>
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+<div class="refentry">
+<a name="GstRTSPSessionPool"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="GstRTSPSessionPool.top_of_page"></a>GstRTSPSessionPool</span></h2>
+<p>GstRTSPSessionPool — An object for managing sessions</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-new" title="gst_rtsp_session_pool_new ()">gst_rtsp_session_pool_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-max-sessions" title="gst_rtsp_session_pool_get_max_sessions ()">gst_rtsp_session_pool_get_max_sessions</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-set-max-sessions" title="gst_rtsp_session_pool_set_max_sessions ()">gst_rtsp_session_pool_set_max_sessions</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-n-sessions" title="gst_rtsp_session_pool_get_n_sessions ()">gst_rtsp_session_pool_get_n_sessions</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create" title="gst_rtsp_session_pool_create ()">gst_rtsp_session_pool_create</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-find" title="gst_rtsp_session_pool_find ()">gst_rtsp_session_pool_find</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-remove" title="gst_rtsp_session_pool_remove ()">gst_rtsp_session_pool_remove</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-cleanup" title="gst_rtsp_session_pool_cleanup ()">gst_rtsp_session_pool_cleanup</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<span class="c_punctuation">(</span><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolFunc" title="GstRTSPSessionPoolFunc ()">*GstRTSPSessionPoolFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="returnvalue">GSource</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create-watch" title="gst_rtsp_session_pool_create_watch ()">gst_rtsp_session_pool_create_watch</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
+</td>
+<td class="function_name">
+<span class="c_punctuation">(</span><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolFilterFunc" title="GstRTSPSessionPoolFilterFunc ()">*GstRTSPSessionPoolFilterFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()">gst_rtsp_session_pool_filter</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.properties"></a><h2>Properties</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="properties_type">
+<col width="300px" class="properties_name">
+<col width="200px" class="properties_flags">
+</colgroup>
+<tbody><tr>
+<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
+<td class="property_name"><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool--max-sessions" title="The “max-sessions” property">max-sessions</a></td>
+<td class="property_flags">Read / Write</td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.signals"></a><h2>Signals</h2>
+<div class="informaltable"><table border="0">
+<colgroup>
+<col width="150px" class="signals_return">
+<col width="300px" class="signals_name">
+<col width="200px" class="signals_flags">
+</colgroup>
+<tbody><tr>
+<td class="signal_type"><span class="returnvalue">void</span></td>
+<td class="signal_name"><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool-session-removed" title="The “session-removed” signal">session-removed</a></td>
+<td class="signal_flags">Run Last</td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool-struct" title="struct GstRTSPSessionPool">GstRTSPSessionPool</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolClass" title="struct GstRTSPSessionPoolClass">GstRTSPSessionPoolClass</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.object-hierarchy"></a><h2>Object Hierarchy</h2>
+<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+ <span class="lineart">╰──</span> GstRTSPSessionPool
+</pre>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.description"></a><h2>Description</h2>
+<p>The <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> object manages a list of <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> objects.</p>
+<p>The maximum number of sessions can be configured with
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-set-max-sessions" title="gst_rtsp_session_pool_set_max_sessions ()"><code class="function">gst_rtsp_session_pool_set_max_sessions()</code></a>. The current number of sessions can
+be retrieved with <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-n-sessions" title="gst_rtsp_session_pool_get_n_sessions ()"><code class="function">gst_rtsp_session_pool_get_n_sessions()</code></a>.</p>
+<p>Use <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create" title="gst_rtsp_session_pool_create ()"><code class="function">gst_rtsp_session_pool_create()</code></a> to create a new <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> object.
+The session object can be found again with its id and
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-find" title="gst_rtsp_session_pool_find ()"><code class="function">gst_rtsp_session_pool_find()</code></a>.</p>
+<p>All sessions can be iterated with <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()"><code class="function">gst_rtsp_session_pool_filter()</code></a>.</p>
+<p>Run <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-cleanup" title="gst_rtsp_session_pool_cleanup ()"><code class="function">gst_rtsp_session_pool_cleanup()</code></a> periodically to remove timed out sessions
+or use <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create-watch" title="gst_rtsp_session_pool_create_watch ()"><code class="function">gst_rtsp_session_pool_create_watch()</code></a> to be notified when session
+cleanup should be performed.</p>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-session-pool-new"></a><h3>gst_rtsp_session_pool_new ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
+gst_rtsp_session_pool_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Create a new <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> instance.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.2.5"></a><h4>Returns</h4>
+<p> A new <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-pool-get-max-sessions"></a><h3>gst_rtsp_session_pool_get_max_sessions ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_session_pool_get_max_sessions
+ (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
+<p>Get the maximum allowed number of sessions in <em class="parameter"><code>pool</code></em>
+. 0 means an unlimited
+amount of sessions.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.9.9.3.6"></a><h4>Returns</h4>
+<p> the maximum allowed number of sessions.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-pool-set-max-sessions"></a><h3>gst_rtsp_session_pool_set_max_sessions ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_session_pool_set_max_sessions
+ (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> max</code></em>);</pre>
+<p>Configure the maximum allowed number of sessions in <em class="parameter"><code>pool</code></em>
+ to <em class="parameter"><code>max</code></em>
+.
+A value of 0 means an unlimited amount of sessions.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>max</p></td>
+<td class="parameter_description"><p>the maximum number of sessions</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-pool-get-n-sessions"></a><h3>gst_rtsp_session_pool_get_n_sessions ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_session_pool_get_n_sessions (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
+<p>Get the amount of active sessions in <em class="parameter"><code>pool</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.9.9.5.6"></a><h4>Returns</h4>
+<p> the amount of active sessions in <em class="parameter"><code>pool</code></em>
+.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-pool-create"></a><h3>gst_rtsp_session_pool_create ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
+gst_rtsp_session_pool_create (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
+<p>Create a new <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> object in <em class="parameter"><code>pool</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.9.9.6.6"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-pool-find"></a><h3>gst_rtsp_session_pool_find ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
+gst_rtsp_session_pool_find (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *sessionid</code></em>);</pre>
+<p>Find the session with <em class="parameter"><code>sessionid</code></em>
+ in <em class="parameter"><code>pool</code></em>
+. The access time of the session
+will be updated with <a class="link" href="GstRTSPSession.html#gst-rtsp-session-touch" title="gst_rtsp_session_touch ()"><code class="function">gst_rtsp_session_touch()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>the pool to search</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>sessionid</p></td>
+<td class="parameter_description"><p>the session id</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.9.9.7.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> with <em class="parameter"><code>sessionid</code></em>
+or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when the session did not exist. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-pool-remove"></a><h3>gst_rtsp_session_pool_remove ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_session_pool_remove (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>);</pre>
+<p>Remove <em class="parameter"><code>sess</code></em>
+ from <em class="parameter"><code>pool</code></em>
+, releasing the ref that the pool has on <em class="parameter"><code>sess</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>sess</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.9.9.8.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the session was found and removed.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-pool-cleanup"></a><h3>gst_rtsp_session_pool_cleanup ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_session_pool_cleanup (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
+<p>Inspect all the sessions in <em class="parameter"><code>pool</code></em>
+ and remove the sessions that are inactive
+for more than their timeout.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.9.9.9.6"></a><h4>Returns</h4>
+<p> the amount of sessions that got removed.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPSessionPoolFunc"></a><h3>GstRTSPSessionPoolFunc ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+<span class="c_punctuation">(</span>*GstRTSPSessionPoolFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>The function that will be called from the GSource watch on the session pool.</p>
+<p>The function will be called when the pool must be cleaned up because one or
+more sessions timed out.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.10.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> object</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data that has been given when registering the handler</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.9.9.10.7"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#FALSE:CAPS"><code class="literal">FALSE</code></a> if the source should be removed.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-pool-create-watch"></a><h3>gst_rtsp_session_pool_create_watch ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="returnvalue">GSource</span></a> *
+gst_rtsp_session_pool_create_watch (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
+<p>Create a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="type">GSource</span></a> that will be dispatched when the session should be cleaned
+up.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.11.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.9.9.11.6"></a><h4>Returns</h4>
+<p> a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="type">GSource</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPSessionPoolFilterFunc"></a><h3>GstRTSPSessionPoolFilterFunc ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
+<span class="c_punctuation">(</span>*GstRTSPSessionPoolFilterFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>This function will be called by the <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()"><code class="function">gst_rtsp_session_pool_filter()</code></a>. An
+implementation should return a value of <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
+<p>When this function returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, <em class="parameter"><code>session</code></em>
+ will be removed
+from <em class="parameter"><code>pool</code></em>
+.</p>
+<p>A return value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a> will leave <em class="parameter"><code>session</code></em>
+ untouched in
+<em class="parameter"><code>pool</code></em>
+.</p>
+<p>A value of GST_RTSP_FILTER_REF will add <em class="parameter"><code>session</code></em>
+ to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()"><code class="function">gst_rtsp_session_pool_filter()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.12.8"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> object</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>session</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> in <em class="parameter"><code>pool</code></em>
+</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data that has been given to <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()"><code class="function">gst_rtsp_session_pool_filter()</code></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.9.9.12.9"></a><h4>Returns</h4>
+<p> a <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-pool-filter"></a><h3>gst_rtsp_session_pool_filter ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
+gst_rtsp_session_pool_filter (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolFilterFunc" title="GstRTSPSessionPoolFilterFunc ()"><span class="type">GstRTSPSessionPoolFilterFunc</span></a> func</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>Call <em class="parameter"><code>func</code></em>
+ for each session in <em class="parameter"><code>pool</code></em>
+. The result value of <em class="parameter"><code>func</code></em>
+ determines
+what happens to the session. <em class="parameter"><code>func</code></em>
+ will be called with the session pool
+locked so no further actions on <em class="parameter"><code>pool</code></em>
+ can be performed from <em class="parameter"><code>func</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, the session will be set to the
+expired state with <code class="function">gst_rtsp_session_set_expired()</code> and removed from
+<em class="parameter"><code>pool</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a>, the session will remain in <em class="parameter"><code>pool</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>, the session will remain in <em class="parameter"><code>pool</code></em>
+ but
+will also be added with an additional ref to the result GList of this
+function..</p>
+<p>When <em class="parameter"><code>func</code></em>
+ is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will be assumed for all sessions.</p>
+<div class="refsect3">
+<a name="id-1.2.9.9.13.9"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>func</p></td>
+<td class="parameter_description"><p> a callback. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid only during the call to the method."><span class="acronym">scope call</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p> user data passed to <em class="parameter"><code>func</code></em>
+. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.9.9.13.10"></a><h4>Returns</h4>
+<p> a GList with all
+sessions for which <em class="parameter"><code>func</code></em>
+returned <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>. After usage, each
+element in the GList should be unreffed before the list is freed. </p>
+<p><span class="annotation">[<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRTSPSession][<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPSessionPool-struct"></a><h3>struct GstRTSPSessionPool</h3>
+<pre class="programlisting">struct GstRTSPSessionPool;</pre>
+<p>An object that keeps track of the active sessions. This object is usually
+attached to a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> object to manage the sessions in that server.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPSessionPoolClass"></a><h3>struct GstRTSPSessionPoolClass</h3>
+<pre class="programlisting">struct GstRTSPSessionPoolClass {
+ GObjectClass parent_class;
+
+ gchar * (*create_session_id) (GstRTSPSessionPool *pool);
+ GstRTSPSession * (*create_session) (GstRTSPSessionPool *pool, const gchar *id);
+
+ /* signals */
+ void (*session_removed) (GstRTSPSessionPool *pool,
+ GstRTSPSession *session);
+};
+</pre>
+<div class="refsect3">
+<a name="id-1.2.9.10.3.4"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPSessionPoolClass.parent-class"></a>parent_class</code></em>;</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPSessionPoolClass.create-session-id"></a>create_session_id</code></em> ()</p></td>
+<td class="struct_member_description"><p>create a new random session id. Subclasses can create
+custom session ids and should not check if the session exists.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPSessionPoolClass.create-session"></a>create_session</code></em> ()</p></td>
+<td class="struct_member_description"><p>make a new session object.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPSessionPoolClass.session-removed"></a>session_removed</code></em> ()</p></td>
+<td class="struct_member_description"><p>a session was removed from the pool</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.property-details"></a><h2>Property Details</h2>
+<div class="refsect2">
+<a name="GstRTSPSessionPool--max-sessions"></a><h3>The <code class="literal">“max-sessions”</code> property</h3>
+<pre class="programlisting"> “max-sessions” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
+<p>the maximum amount of sessions (0 = unlimited).</p>
+<p>Flags: Read / Write</p>
+<p>Default value: 0</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.signal-details"></a><h2>Signal Details</h2>
+<div class="refsect2">
+<a name="GstRTSPSessionPool-session-removed"></a><h3>The <code class="literal">“session-removed”</code> signal</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+user_function (<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *gstrtspsessionpool,
+ <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *arg1,
+ <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
+<p>Flags: Run Last</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPSessionPool.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
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+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
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+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#GstRTSPStream.description" class="shortcut">Description</a></span><span id="nav_hierarchy"> <span class="dim">|</span>
+ <a href="#GstRTSPStream.object-hierarchy" class="shortcut">Object Hierarchy</a></span>
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+<div class="refentry">
+<a name="GstRTSPStream"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="GstRTSPStream.top_of_page"></a>GstRTSPStream</span></h2>
+<p>GstRTSPStream — A media stream</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="GstRTSPStream.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-new" title="gst_rtsp_stream_new ()">gst_rtsp_stream_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-index" title="gst_rtsp_stream_get_index ()">gst_rtsp_stream_get_index</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="returnvalue">GstPad</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-srcpad" title="gst_rtsp_stream_get_srcpad ()">gst_rtsp_stream_get_srcpad</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-control" title="gst_rtsp_stream_get_control ()">gst_rtsp_stream_get_control</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-control" title="gst_rtsp_stream_set_control ()">gst_rtsp_stream_set_control</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-has-control" title="gst_rtsp_stream_has_control ()">gst_rtsp_stream_has_control</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-mtu" title="gst_rtsp_stream_get_mtu ()">gst_rtsp_stream_get_mtu</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-mtu" title="gst_rtsp_stream_set_mtu ()">gst_rtsp_stream_set_mtu</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-dscp-qos" title="gst_rtsp_stream_get_dscp_qos ()">gst_rtsp_stream_get_dscp_qos</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-dscp-qos" title="gst_rtsp_stream_set_dscp_qos ()">gst_rtsp_stream_set_dscp_qos</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-profiles" title="gst_rtsp_stream_set_profiles ()">gst_rtsp_stream_set_profiles</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-profiles" title="gst_rtsp_stream_get_profiles ()">gst_rtsp_stream_get_profiles</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-protocols" title="gst_rtsp_stream_get_protocols ()">gst_rtsp_stream_get_protocols</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-protocols" title="gst_rtsp_stream_set_protocols ()">gst_rtsp_stream_set_protocols</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-is-transport-supported" title="gst_rtsp_stream_is_transport_supported ()">gst_rtsp_stream_is_transport_supported</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-address-pool" title="gst_rtsp_stream_get_address_pool ()">gst_rtsp_stream_get_address_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-address-pool" title="gst_rtsp_stream_set_address_pool ()">gst_rtsp_stream_set_address_pool</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-reserve-address" title="gst_rtsp_stream_reserve_address ()">gst_rtsp_stream_reserve_address</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-join-bin" title="gst_rtsp_stream_join_bin ()">gst_rtsp_stream_join_bin</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-leave-bin" title="gst_rtsp_stream_leave_bin ()">gst_rtsp_stream_leave_bin</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-server-port" title="gst_rtsp_stream_get_server_port ()">gst_rtsp_stream_get_server_port</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-multicast-address" title="gst_rtsp_stream_get_multicast_address ()">gst_rtsp_stream_get_multicast_address</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject"><span class="returnvalue">GObject</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtpsession" title="gst_rtsp_stream_get_rtpsession ()">gst_rtsp_stream_get_rtpsession</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-ssrc" title="gst_rtsp_stream_get_ssrc ()">gst_rtsp_stream_get_ssrc</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtpinfo" title="gst_rtsp_stream_get_rtpinfo ()">gst_rtsp_stream_get_rtpinfo</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html"><span class="returnvalue">GstCaps</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-caps" title="gst_rtsp_stream_get_caps ()">gst_rtsp_stream_get_caps</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-pt" title="gst_rtsp_stream_get_pt ()">gst_rtsp_stream_get_pt</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-recv-rtcp" title="gst_rtsp_stream_recv_rtcp ()">gst_rtsp_stream_recv_rtcp</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-recv-rtp" title="gst_rtsp_stream_recv_rtp ()">gst_rtsp_stream_recv_rtp</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-add-transport" title="gst_rtsp_stream_add_transport ()">gst_rtsp_stream_add_transport</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-remove-transport" title="gst_rtsp_stream_remove_transport ()">gst_rtsp_stream_remove_transport</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtp-socket" title="gst_rtsp_stream_get_rtp_socket ()">gst_rtsp_stream_get_rtp_socket</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtcp-socket" title="gst_rtsp_stream_get_rtcp_socket ()">gst_rtsp_stream_get_rtcp_socket</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-blocked" title="gst_rtsp_stream_set_blocked ()">gst_rtsp_stream_set_blocked</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-is-blocking" title="gst_rtsp_stream_is_blocking ()">gst_rtsp_stream_is_blocking</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-update-crypto" title="gst_rtsp_stream_update_crypto ()">gst_rtsp_stream_update_crypto</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
+</td>
+<td class="function_name">
+<span class="c_punctuation">(</span><a class="link" href="GstRTSPStream.html#GstRTSPStreamTransportFilterFunc" title="GstRTSPStreamTransportFilterFunc ()">*GstRTSPStreamTransportFilterFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-transport-filter" title="gst_rtsp_stream_transport_filter ()">gst_rtsp_stream_transport_filter</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPStream.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="GstRTSPStream.html#GstRTSPStream-struct" title="struct GstRTSPStream">GstRTSPStream</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword"> </td>
+<td class="function_name"><a class="link" href="GstRTSPStream.html#GstRTSPStreamClass" title="GstRTSPStreamClass">GstRTSPStreamClass</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPStream.object-hierarchy"></a><h2>Object Hierarchy</h2>
+<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+ <span class="lineart">╰──</span> GstRTSPStream
+</pre>
+</div>
+<div class="refsect1">
+<a name="GstRTSPStream.description"></a><h2>Description</h2>
+<p>The <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> object manages the data transport for one stream. It
+is created from a payloader element and a source pad that produce the RTP
+packets for the stream.</p>
+<p>With <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-join-bin" title="gst_rtsp_stream_join_bin ()"><code class="function">gst_rtsp_stream_join_bin()</code></a> the streaming elements are added to the bin
+and rtpbin. <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-leave-bin" title="gst_rtsp_stream_leave_bin ()"><code class="function">gst_rtsp_stream_leave_bin()</code></a> removes the elements again.</p>
+<p>The <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> will use the configured addresspool, as set with
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-address-pool" title="gst_rtsp_stream_set_address_pool ()"><code class="function">gst_rtsp_stream_set_address_pool()</code></a>, to allocate multicast addresses for the
+stream. With <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-multicast-address" title="gst_rtsp_stream_get_multicast_address ()"><code class="function">gst_rtsp_stream_get_multicast_address()</code></a> you can get the
+configured address.</p>
+<p>With <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-server-port" title="gst_rtsp_stream_get_server_port ()"><code class="function">gst_rtsp_stream_get_server_port()</code></a> you can get the port that the server
+will use to receive RTCP. This is the part that the clients will use to send
+RTCP to.</p>
+<p>With <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-add-transport" title="gst_rtsp_stream_add_transport ()"><code class="function">gst_rtsp_stream_add_transport()</code></a> destinations can be added where the
+stream should be sent to. Use <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-remove-transport" title="gst_rtsp_stream_remove_transport ()"><code class="function">gst_rtsp_stream_remove_transport()</code></a> to remove
+the destination again.</p>
+<p>Last reviewed on 2013-07-16 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="GstRTSPStream.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-stream-new"></a><h3>gst_rtsp_stream_new ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
+gst_rtsp_stream_new (<em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> idx</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *payloader</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="type">GstPad</span></a> *srcpad</code></em>);</pre>
+<p>Create a new media stream with index <em class="parameter"><code>idx</code></em>
+ that handles RTP data on
+<em class="parameter"><code>srcpad</code></em>
+ and has a payloader element <em class="parameter"><code>payloader</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.2.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>idx</p></td>
+<td class="parameter_description"><p>an index</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>srcpad</p></td>
+<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="type">GstPad</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>payloader</p></td>
+<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.2.6"></a><h4>Returns</h4>
+<p> a new <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-index"></a><h3>gst_rtsp_stream_get_index ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_stream_get_index (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Get the stream index.</p>
+<p>Return: the stream index.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.3.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-srcpad"></a><h3>gst_rtsp_stream_get_srcpad ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="returnvalue">GstPad</span></a> *
+gst_rtsp_stream_get_srcpad (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Get the srcpad associated with <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.4.6"></a><h4>Returns</h4>
+<p> the srcpad. Unref after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-control"></a><h3>gst_rtsp_stream_get_control ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_stream_get_control (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Get the control string to identify this stream.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.5.6"></a><h4>Returns</h4>
+<p> the control string. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-set-control"></a><h3>gst_rtsp_stream_set_control ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_set_control (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *control</code></em>);</pre>
+<p>Set the control string in <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>control</p></td>
+<td class="parameter_description"><p>a control string</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-has-control"></a><h3>gst_rtsp_stream_has_control ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_has_control (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *control</code></em>);</pre>
+<p>Check if <em class="parameter"><code>stream</code></em>
+ has the control string <em class="parameter"><code>control</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>control</p></td>
+<td class="parameter_description"><p>a control string</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.7.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> is <em class="parameter"><code>stream</code></em>
+has <em class="parameter"><code>control</code></em>
+as the control string</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-mtu"></a><h3>gst_rtsp_stream_get_mtu ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_stream_get_mtu (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Get the configured MTU in the payloader of <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.8.6"></a><h4>Returns</h4>
+<p> the MTU of the payloader.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-set-mtu"></a><h3>gst_rtsp_stream_set_mtu ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_set_mtu (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> mtu</code></em>);</pre>
+<p>Configure the mtu in the payloader of <em class="parameter"><code>stream</code></em>
+ to <em class="parameter"><code>mtu</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>mtu</p></td>
+<td class="parameter_description"><p>a new MTU</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-dscp-qos"></a><h3>gst_rtsp_stream_get_dscp_qos ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
+gst_rtsp_stream_get_dscp_qos (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Get the configured DSCP QoS in of the outgoing sockets.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.10.6"></a><h4>Returns</h4>
+<p> the DSCP QoS value of the outgoing sockets, or -1 if disbled.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-set-dscp-qos"></a><h3>gst_rtsp_stream_set_dscp_qos ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_set_dscp_qos (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> dscp_qos</code></em>);</pre>
+<p>Configure the dscp qos of the outgoing sockets to <em class="parameter"><code>dscp_qos</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.11.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>dscp_qos</p></td>
+<td class="parameter_description"><p>a new dscp qos value (0-63, or -1 to disable)</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-set-profiles"></a><h3>gst_rtsp_stream_set_profiles ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_set_profiles (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a> profiles</code></em>);</pre>
+<p>Configure the allowed profiles for <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.12.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>profiles</p></td>
+<td class="parameter_description"><p>the new profiles</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-profiles"></a><h3>gst_rtsp_stream_get_profiles ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
+gst_rtsp_stream_get_profiles (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Get the allowed profiles of <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.13.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.13.6"></a><h4>Returns</h4>
+<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-protocols"></a><h3>gst_rtsp_stream_get_protocols ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
+gst_rtsp_stream_get_protocols (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Get the allowed protocols of <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.14.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.14.6"></a><h4>Returns</h4>
+<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-set-protocols"></a><h3>gst_rtsp_stream_set_protocols ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_set_protocols (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a> protocols</code></em>);</pre>
+<p>Configure the allowed lower transport for <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.15.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>protocols</p></td>
+<td class="parameter_description"><p>the new flags</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-is-transport-supported"></a><h3>gst_rtsp_stream_is_transport_supported ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_is_transport_supported
+ (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a> *transport</code></em>);</pre>
+<p>Check if <em class="parameter"><code>transport</code></em>
+ can be handled by stream</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.16.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>transport</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.16.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>transport</code></em>
+can be handled by <em class="parameter"><code>stream</code></em>
+.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-address-pool"></a><h3>gst_rtsp_stream_get_address_pool ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
+gst_rtsp_stream_get_address_pool (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Get the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> used as the address pool of <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.17.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.17.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> of <em class="parameter"><code>stream</code></em>
+. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
+usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-set-address-pool"></a><h3>gst_rtsp_stream_set_address_pool ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_set_address_pool (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
+<p>configure <em class="parameter"><code>pool</code></em>
+ to be used as the address pool of <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.18.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-reserve-address"></a><h3>gst_rtsp_stream_reserve_address ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
+gst_rtsp_stream_reserve_address (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *address</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> port</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> n_ports</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> ttl</code></em>);</pre>
+<p>Reserve <em class="parameter"><code>address</code></em>
+ and <em class="parameter"><code>port</code></em>
+ as the address and port of <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.19.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>address</p></td>
+<td class="parameter_description"><p>an address</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>port</p></td>
+<td class="parameter_description"><p>a port</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>n_ports</p></td>
+<td class="parameter_description"><p>n_ports</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>ttl</p></td>
+<td class="parameter_description"><p>a TTL</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.19.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> of <em class="parameter"><code>stream</code></em>
+or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when
+the address could be reserved. <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free" title="gst_rtsp_address_free ()"><code class="function">gst_rtsp_address_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-join-bin"></a><h3>gst_rtsp_stream_join_bin ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_join_bin (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> *bin</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *rtpbin</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GstState"><span class="type">GstState</span></a> state</code></em>);</pre>
+<p>Join the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> <em class="parameter"><code>bin</code></em>
+ that contains the element <em class="parameter"><code>rtpbin</code></em>
+.</p>
+<p><em class="parameter"><code>stream</code></em>
+ will link to <em class="parameter"><code>rtpbin</code></em>
+, which must be inside <em class="parameter"><code>bin</code></em>
+. The elements
+added to <em class="parameter"><code>bin</code></em>
+ will be set to the state given in <em class="parameter"><code>state</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.20.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>bin</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> to join. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>rtpbin</p></td>
+<td class="parameter_description"><p> a rtpbin element in <em class="parameter"><code>bin</code></em>
+. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>state</p></td>
+<td class="parameter_description"><p>the target state of the new elements</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.20.7"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-leave-bin"></a><h3>gst_rtsp_stream_leave_bin ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_leave_bin (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> *bin</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *rtpbin</code></em>);</pre>
+<p>Remove the elements of <em class="parameter"><code>stream</code></em>
+ from <em class="parameter"><code>bin</code></em>
+.</p>
+<p>Return: <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.21.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>bin</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>rtpbin</p></td>
+<td class="parameter_description"><p> a rtpbin <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-server-port"></a><h3>gst_rtsp_stream_get_server_port ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_get_server_port (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPRange"><span class="type">GstRTSPRange</span></a> *server_port</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocketAddress.html#GSocketFamily"><span class="type">GSocketFamily</span></a> family</code></em>);</pre>
+<p>Fill <em class="parameter"><code>server_port</code></em>
+ with the port pair used by the server. This function can
+only be called when <em class="parameter"><code>stream</code></em>
+ has been joined.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.22.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>server_port</p></td>
+<td class="parameter_description"><p> result server port. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>family</p></td>
+<td class="parameter_description"><p>the port family to get</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-multicast-address"></a><h3>gst_rtsp_stream_get_multicast_address ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
+gst_rtsp_stream_get_multicast_address (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocketAddress.html#GSocketFamily"><span class="type">GSocketFamily</span></a> family</code></em>);</pre>
+<p>Get the multicast address of <em class="parameter"><code>stream</code></em>
+ for <em class="parameter"><code>family</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.23.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>family</p></td>
+<td class="parameter_description"><p>the <a href="https://developer.gnome.org/gio/unstable/GSocketAddress.html#GSocketFamily"><span class="type">GSocketFamily</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.23.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> of <em class="parameter"><code>stream</code></em>
+or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when no address could be allocated. <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free" title="gst_rtsp_address_free ()"><code class="function">gst_rtsp_address_free()</code></a>
+after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-rtpsession"></a><h3>gst_rtsp_stream_get_rtpsession ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject"><span class="returnvalue">GObject</span></a> *
+gst_rtsp_stream_get_rtpsession (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Get the RTP session of this stream.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.24.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.24.6"></a><h4>Returns</h4>
+<p> The RTP session of this stream. Unref after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-ssrc"></a><h3>gst_rtsp_stream_get_ssrc ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_get_ssrc (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> *ssrc</code></em>);</pre>
+<p>Get the SSRC used by the RTP session of this stream. This function can only
+be called when <em class="parameter"><code>stream</code></em>
+ has been joined.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.25.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>ssrc</p></td>
+<td class="parameter_description"><p> result ssrc. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-rtpinfo"></a><h3>gst_rtsp_stream_get_rtpinfo ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_get_rtpinfo (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> *rtptime</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> *seq</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> *clock_rate</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a> *running_time</code></em>);</pre>
+<p>Retrieve the current rtptime, seq and running-time. This is used to
+construct a RTPInfo reply header.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.26.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>rtptime</p></td>
+<td class="parameter_description"><p> result RTP timestamp. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>seq</p></td>
+<td class="parameter_description"><p> result RTP seqnum. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>clock_rate</p></td>
+<td class="parameter_description"><p> the clock rate. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>running_time</p></td>
+<td class="parameter_description"><p> result running-time. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.26.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> when rtptime, seq and running-time could be determined.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-caps"></a><h3>gst_rtsp_stream_get_caps ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html"><span class="returnvalue">GstCaps</span></a> *
+gst_rtsp_stream_get_caps (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Retrieve the current caps of <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.27.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.27.6"></a><h4>Returns</h4>
+<p> the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html"><span class="type">GstCaps</span></a> of <em class="parameter"><code>stream</code></em>
+. use <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html#gst-caps-unref"><code class="function">gst_caps_unref()</code></a>
+after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-pt"></a><h3>gst_rtsp_stream_get_pt ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
+gst_rtsp_stream_get_pt (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Get the stream payload type.</p>
+<p>Return: the stream payload type.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.28.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-recv-rtcp"></a><h3>gst_rtsp_stream_recv_rtcp ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>
+gst_rtsp_stream_recv_rtcp (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a> *buffer</code></em>);</pre>
+<p>Handle an RTCP buffer for the stream. This method is usually called when a
+message has been received from a client using the TCP transport.</p>
+<p>This function takes ownership of <em class="parameter"><code>buffer</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.29.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>buffer</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.29.7"></a><h4>Returns</h4>
+<p> a GstFlowReturn.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-recv-rtp"></a><h3>gst_rtsp_stream_recv_rtp ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>
+gst_rtsp_stream_recv_rtp (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a> *buffer</code></em>);</pre>
+<p>Handle an RTP buffer for the stream. This method is usually called when a
+message has been received from a client using the TCP transport.</p>
+<p>This function takes ownership of <em class="parameter"><code>buffer</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.30.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>buffer</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.30.7"></a><h4>Returns</h4>
+<p> a GstFlowReturn.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-add-transport"></a><h3>gst_rtsp_stream_add_transport ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_add_transport (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
+<p>Add the transport in <em class="parameter"><code>trans</code></em>
+ to <em class="parameter"><code>stream</code></em>
+. The media of <em class="parameter"><code>stream</code></em>
+ will
+then also be send to the values configured in <em class="parameter"><code>trans</code></em>
+.</p>
+<p><em class="parameter"><code>stream</code></em>
+ must be joined to a bin.</p>
+<p><em class="parameter"><code>trans</code></em>
+ must contain a valid <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.31.7"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.31.8"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>trans</code></em>
+was added</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-remove-transport"></a><h3>gst_rtsp_stream_remove_transport ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_remove_transport (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
+<p>Remove the transport in <em class="parameter"><code>trans</code></em>
+ from <em class="parameter"><code>stream</code></em>
+. The media of <em class="parameter"><code>stream</code></em>
+ will
+not be sent to the values configured in <em class="parameter"><code>trans</code></em>
+.</p>
+<p><em class="parameter"><code>stream</code></em>
+ must be joined to a bin.</p>
+<p><em class="parameter"><code>trans</code></em>
+ must contain a valid <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.32.7"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.32.8"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>trans</code></em>
+was removed</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-rtp-socket"></a><h3>gst_rtsp_stream_get_rtp_socket ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
+gst_rtsp_stream_get_rtp_socket (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocketAddress.html#GSocketFamily"><span class="type">GSocketFamily</span></a> family</code></em>);</pre>
+<p>Get the RTP socket from <em class="parameter"><code>stream</code></em>
+ for a <em class="parameter"><code>family</code></em>
+.</p>
+<p><em class="parameter"><code>stream</code></em>
+ must be joined to a bin.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.33.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>family</p></td>
+<td class="parameter_description"><p>the socket family</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.33.7"></a><h4>Returns</h4>
+<p> the RTP socket or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> if no
+socket could be allocated for <em class="parameter"><code>family</code></em>
+. Unref after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-get-rtcp-socket"></a><h3>gst_rtsp_stream_get_rtcp_socket ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
+gst_rtsp_stream_get_rtcp_socket (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocketAddress.html#GSocketFamily"><span class="type">GSocketFamily</span></a> family</code></em>);</pre>
+<p>Get the RTCP socket from <em class="parameter"><code>stream</code></em>
+ for a <em class="parameter"><code>family</code></em>
+.</p>
+<p><em class="parameter"><code>stream</code></em>
+ must be joined to a bin.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.34.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>family</p></td>
+<td class="parameter_description"><p>the socket family</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.34.7"></a><h4>Returns</h4>
+<p> the RTCP socket or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> if no
+socket could be allocated for <em class="parameter"><code>family</code></em>
+. Unref after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-set-blocked"></a><h3>gst_rtsp_stream_set_blocked ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_set_blocked (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> blocked</code></em>);</pre>
+<p>Blocks or unblocks the dataflow on <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.35.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>blocked</p></td>
+<td class="parameter_description"><p>boolean indicating we should block or unblock</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.35.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-is-blocking"></a><h3>gst_rtsp_stream_is_blocking ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_is_blocking (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
+<p>Check if <em class="parameter"><code>stream</code></em>
+ is blocking on a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.36.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.36.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>stream</code></em>
+is blocking</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-update-crypto"></a><h3>gst_rtsp_stream_update_crypto ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_update_crypto (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> ssrc</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html"><span class="type">GstCaps</span></a> *crypto</code></em>);</pre>
+<p>Update the new crypto information for <em class="parameter"><code>ssrc</code></em>
+ in <em class="parameter"><code>stream</code></em>
+. If information
+for <em class="parameter"><code>ssrc</code></em>
+ did not exist, it will be added. If information
+for <em class="parameter"><code>ssrc</code></em>
+ existed, it will be replaced. If <em class="parameter"><code>crypto</code></em>
+ is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, it will
+be removed from <em class="parameter"><code>stream</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.37.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>ssrc</p></td>
+<td class="parameter_description"><p>the SSRC</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>crypto</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html"><span class="type">GstCaps</span></a> with crypto info. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.37.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>crypto</code></em>
+could be updated</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPStreamTransportFilterFunc"></a><h3>GstRTSPStreamTransportFilterFunc ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
+<span class="c_punctuation">(</span>*GstRTSPStreamTransportFilterFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>This function will be called by the <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-transport-filter" title="gst_rtsp_stream_transport_filter ()"><code class="function">gst_rtsp_stream_transport_filter()</code></a>. An
+implementation should return a value of <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
+<p>When this function returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, <em class="parameter"><code>trans</code></em>
+ will be removed
+from <em class="parameter"><code>stream</code></em>
+.</p>
+<p>A return value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a> will leave <em class="parameter"><code>trans</code></em>
+ untouched in
+<em class="parameter"><code>stream</code></em>
+.</p>
+<p>A value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will add <em class="parameter"><code>trans</code></em>
+ to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-transport-filter" title="gst_rtsp_stream_transport_filter ()"><code class="function">gst_rtsp_stream_transport_filter()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.38.8"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> object</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> in <em class="parameter"><code>stream</code></em>
+</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data that has been given to <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-transport-filter" title="gst_rtsp_stream_transport_filter ()"><code class="function">gst_rtsp_stream_transport_filter()</code></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.38.9"></a><h4>Returns</h4>
+<p> a <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-filter"></a><h3>gst_rtsp_stream_transport_filter ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
+gst_rtsp_stream_transport_filter (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPStream.html#GstRTSPStreamTransportFilterFunc" title="GstRTSPStreamTransportFilterFunc ()"><span class="type">GstRTSPStreamTransportFilterFunc</span></a> func</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>Call <em class="parameter"><code>func</code></em>
+ for each transport managed by <em class="parameter"><code>stream</code></em>
+. The result value of <em class="parameter"><code>func</code></em>
+
+determines what happens to the transport. <em class="parameter"><code>func</code></em>
+ will be called with <em class="parameter"><code>stream</code></em>
+
+locked so no further actions on <em class="parameter"><code>stream</code></em>
+ can be performed from <em class="parameter"><code>func</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, the transport will be removed from
+<em class="parameter"><code>stream</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a>, the transport will remain in <em class="parameter"><code>stream</code></em>
+.</p>
+<p>If <em class="parameter"><code>func</code></em>
+ returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>, the transport will remain in <em class="parameter"><code>stream</code></em>
+ but
+will also be added with an additional ref to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of this
+function..</p>
+<p>When <em class="parameter"><code>func</code></em>
+ is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will be assumed for each transport.</p>
+<div class="refsect3">
+<a name="id-1.2.8.7.39.9"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>func</p></td>
+<td class="parameter_description"><p> a callback. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid only during the call to the method."><span class="acronym">scope call</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p> user data passed to <em class="parameter"><code>func</code></em>
+. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.8.7.39.10"></a><h4>Returns</h4>
+<p> a <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> with all
+transports for which <em class="parameter"><code>func</code></em>
+returned <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>. After usage, each
+element in the <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> should be unreffed before the list is freed. </p>
+<p><span class="annotation">[<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRTSPStreamTransport][<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPStream.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPStream-struct"></a><h3>struct GstRTSPStream</h3>
+<pre class="programlisting">struct GstRTSPStream;</pre>
+<p>The definition of a media stream.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPStreamClass"></a><h3>GstRTSPStreamClass</h3>
+<pre class="programlisting">typedef struct _GstRTSPStreamClass GstRTSPStreamClass;</pre>
+</div>
+</div>
+<div class="refsect1">
+<a name="GstRTSPStream.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
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+ <span class="dim">|</span>
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+ <a class="shortcut" href="#glsE">E</a>
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+<div class="glossary">
+<div class="titlepage"><div><div><h1 class="title">
+<a name="annotation-glossary"></a>Annotation Glossary</h1></div></div></div>
+<a name="glsA"></a><h3 class="title">A</h3>
+<dt><span class="glossterm"><a name="annotation-glossterm-allow-none"></a>allow-none</span></dt>
+<dd class="glossdef"><p>NULL is OK, both for passing and for returning.</p></dd>
+<a name="glsC"></a><h3 class="title">C</h3>
+<dt><span class="glossterm"><a name="annotation-glossterm-closure"></a>closure</span></dt>
+<dd class="glossdef"><p>This parameter is a 'user_data', for callbacks; many bindings can pass NULL here.</p></dd>
+<a name="glsE"></a><h3 class="title">E</h3>
+<dt><span class="glossterm"><a name="annotation-glossterm-element-type"></a>element-type</span></dt>
+<dd class="glossdef"><p>Generics and defining elements of containers and arrays.</p></dd>
+<a name="glsN"></a><h3 class="title">N</h3>
+<dt><span class="glossterm"><a name="annotation-glossterm-nullable"></a>nullable</span></dt>
+<dd class="glossdef"><p>NULL may be passed as the value in, out, in-out; or as a return value.</p></dd>
+<a name="glsO"></a><h3 class="title">O</h3>
+<dt><span class="glossterm"><a name="annotation-glossterm-out"></a>out</span></dt>
+<dd class="glossdef"><p>Parameter for returning results. Default is <acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>.</p></dd>
+<a name="glsS"></a><h3 class="title">S</h3>
+<dt><span class="glossterm"><a name="annotation-glossterm-scope%20call"></a>scope call</span></dt>
+<dd class="glossdef"><p>The callback is valid only during the call to the method.</p></dd>
+<dt><span class="glossterm"><a name="annotation-glossterm-scope%20notified"></a>scope notified</span></dt>
+<dd class="glossdef"><p>The callback is valid until the GDestroyNotify argument is called.</p></dd>
+<a name="glsT"></a><h3 class="title">T</h3>
+<dt><span class="glossterm"><a name="annotation-glossterm-transfer%20floating"></a>transfer floating</span></dt>
+<dd class="glossdef"><p>Alias for <acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>, used for objects with floating refs.</p></dd>
+<dt><span class="glossterm"><a name="annotation-glossterm-transfer%20full"></a>transfer full</span></dt>
+<dd class="glossdef"><p>Free data after the code is done.</p></dd>
+<dt><span class="glossterm"><a name="annotation-glossterm-transfer%20none"></a>transfer none</span></dt>
+<dd class="glossdef"><p>Don't free data after the code is done.</p></dd>
+<dt><span class="glossterm"><a name="annotation-glossterm-type"></a>type</span></dt>
+<dd class="glossdef"><p>Override the parsed C type with given type.</p></dd>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
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+<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
+<title>GStreamer RTSP Server Reference Manual: API Index</title>
+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
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+<td width="100%" align="left" class="shortcuts"><span id="nav_index"><a class="shortcut" href="#idxR">R</a>
+ <span class="dim">|</span>
+ <a class="shortcut" href="#idxS">S</a></span></td>
+<td><a accesskey="h" href="index.html"><img src="home.png" width="16" height="16" border="0" alt="Home"></a></td>
+<td><img src="up-insensitive.png" width="16" height="16" border="0"></td>
+<td><a accesskey="p" href="rtsp-server-hierarchy.html"><img src="left.png" width="16" height="16" border="0" alt="Prev"></a></td>
+<td><a accesskey="n" href="annotation-glossary.html"><img src="right.png" width="16" height="16" border="0" alt="Next"></a></td>
+</tr></table>
+<div class="index">
+<div class="titlepage"><div><div><h1 class="title">
+<a name="api-index-full"></a>API Index</h1></div></div></div>
+<a name="idx"></a><a name="idxR"></a><h3 class="title">R</h3>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress">GstRTSPAddress</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressFlags" title="enum GstRTSPAddressFlags">GstRTSPAddressFlags</a>, enum in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool">GstRTSPAddressPool</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolClass" title="struct GstRTSPAddressPoolClass">GstRTSPAddressPoolClass</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolResult" title="enum GstRTSPAddressPoolResult">GstRTSPAddressPoolResult</a>, enum in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GstRTSPAuth-struct" title="struct GstRTSPAuth">GstRTSPAuth</a>, struct in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GstRTSPAuthClass" title="struct GstRTSPAuthClass">GstRTSPAuthClass</a>, struct in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-struct" title="struct GstRTSPClient">GstRTSPClient</a>, struct in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-closed" title="The “closed” signal">GstRTSPClient::closed</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-describe-request" title="The “describe-request” signal">GstRTSPClient::describe-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-get-parameter-request" title="The “get-parameter-request” signal">GstRTSPClient::get-parameter-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-handle-response" title="The “handle-response” signal">GstRTSPClient::handle-response</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-new-session" title="The “new-session” signal">GstRTSPClient::new-session</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-options-request" title="The “options-request” signal">GstRTSPClient::options-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-pause-request" title="The “pause-request” signal">GstRTSPClient::pause-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-play-request" title="The “play-request” signal">GstRTSPClient::play-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-send-message" title="The “send-message” signal">GstRTSPClient::send-message</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-set-parameter-request" title="The “set-parameter-request” signal">GstRTSPClient::set-parameter-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-setup-request" title="The “setup-request” signal">GstRTSPClient::setup-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient-teardown-request" title="The “teardown-request” signal">GstRTSPClient::teardown-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient--drop-backlog" title="The “drop-backlog” property">GstRTSPClient:drop-backlog</a>, object property in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient--mount-points" title="The “mount-points” property">GstRTSPClient:mount-points</a>, object property in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClient--session-pool" title="The “session-pool” property">GstRTSPClient:session-pool</a>, object property in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClientClass" title="struct GstRTSPClientClass">GstRTSPClientClass</a>, struct in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClientSendFunc" title="GstRTSPClientSendFunc ()">GstRTSPClientSendFunc</a>, user_function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#GstRTSPClientSessionFilterFunc" title="GstRTSPClientSessionFilterFunc ()">GstRTSPClientSessionFilterFunc</a>, user_function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext">GstRTSPContext</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPContext.html" title="GstRTSPContext">GstRTSPContext</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult">GstRTSPFilterResult</a>, enum in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPKeepAliveFunc" title="GstRTSPKeepAliveFunc ()">GstRTSPKeepAliveFunc</a>, user_function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-struct" title="struct GstRTSPMedia">GstRTSPMedia</a>, struct in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-new-state" title="The “new-state” signal">GstRTSPMedia::new-state</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-new-stream" title="The “new-stream” signal">GstRTSPMedia::new-stream</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-prepared" title="The “prepared” signal">GstRTSPMedia::prepared</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-removed-stream" title="The “removed-stream” signal">GstRTSPMedia::removed-stream</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-target-state" title="The “target-state” signal">GstRTSPMedia::target-state</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-unprepared" title="The “unprepared” signal">GstRTSPMedia::unprepared</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--buffer-size" title="The “buffer-size” property">GstRTSPMedia:buffer-size</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--element" title="The “element” property">GstRTSPMedia:element</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--eos-shutdown" title="The “eos-shutdown” property">GstRTSPMedia:eos-shutdown</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--profiles" title="The “profiles” property">GstRTSPMedia:profiles</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--protocols" title="The “protocols” property">GstRTSPMedia:protocols</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--reusable" title="The “reusable” property">GstRTSPMedia:reusable</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--shared" title="The “shared” property">GstRTSPMedia:shared</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--suspend-mode" title="The “suspend-mode” property">GstRTSPMedia:suspend-mode</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--time-provider" title="The “time-provider” property">GstRTSPMedia:time-provider</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMediaClass" title="struct GstRTSPMediaClass">GstRTSPMediaClass</a>, struct in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-struct" title="struct GstRTSPMediaFactory">GstRTSPMediaFactory</a>, struct in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-configure" title="The “media-configure” signal">GstRTSPMediaFactory::media-configure</a>, object signal in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-constructed" title="The “media-constructed” signal">GstRTSPMediaFactory::media-constructed</a>, object signal in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--buffer-size" title="The “buffer-size” property">GstRTSPMediaFactory:buffer-size</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--eos-shutdown" title="The “eos-shutdown” property">GstRTSPMediaFactory:eos-shutdown</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--launch" title="The “launch” property">GstRTSPMediaFactory:launch</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--profiles" title="The “profiles” property">GstRTSPMediaFactory:profiles</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--protocols" title="The “protocols” property">GstRTSPMediaFactory:protocols</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--shared" title="The “shared” property">GstRTSPMediaFactory:shared</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--suspend-mode" title="The “suspend-mode” property">GstRTSPMediaFactory:suspend-mode</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactoryClass" title="struct GstRTSPMediaFactoryClass">GstRTSPMediaFactoryClass</a>, struct in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURIClass" title="struct GstRTSPMediaFactoryURIClass">GstRTSPMediaFactoryURIClass</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPMediaStatus" title="enum GstRTSPMediaStatus">GstRTSPMediaStatus</a>, enum in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMountPoints.html#GstRTSPMountPoints-struct" title="struct GstRTSPMountPoints">GstRTSPMountPoints</a>, struct in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMountPoints.html#GstRTSPMountPointsClass" title="struct GstRTSPMountPointsClass">GstRTSPMountPointsClass</a>, struct in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions">GstRTSPPermissions</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPSendFunc" title="GstRTSPSendFunc ()">GstRTSPSendFunc</a>, user_function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#GstRTSPServer-struct" title="struct GstRTSPServer">GstRTSPServer</a>, struct in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#GstRTSPServer-client-connected" title="The “client-connected” signal">GstRTSPServer::client-connected</a>, object signal in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#GstRTSPServer--address" title="The “address” property">GstRTSPServer:address</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#GstRTSPServer--backlog" title="The “backlog” property">GstRTSPServer:backlog</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#GstRTSPServer--bound-port" title="The “bound-port” property">GstRTSPServer:bound-port</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#GstRTSPServer--mount-points" title="The “mount-points” property">GstRTSPServer:mount-points</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#GstRTSPServer--service" title="The “service” property">GstRTSPServer:service</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#GstRTSPServer--session-pool" title="The “session-pool” property">GstRTSPServer:session-pool</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#GstRTSPServerClass" title="struct GstRTSPServerClass">GstRTSPServerClass</a>, struct in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#GstRTSPServerClientFilterFunc" title="GstRTSPServerClientFilterFunc ()">GstRTSPServerClientFilterFunc</a>, user_function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#GstRTSPSession-struct" title="struct GstRTSPSession">GstRTSPSession</a>, struct in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#GstRTSPSession--sessionid" title="The “sessionid” property">GstRTSPSession:sessionid</a>, object property in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#GstRTSPSession--timeout" title="The “timeout” property">GstRTSPSession:timeout</a>, object property in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#GstRTSPSession--timeout-always-visible" title="The “timeout-always-visible” property">GstRTSPSession:timeout-always-visible</a>, object property in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#GstRTSPSessionClass" title="struct GstRTSPSessionClass">GstRTSPSessionClass</a>, struct in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#GstRTSPSessionFilterFunc" title="GstRTSPSessionFilterFunc ()">GstRTSPSessionFilterFunc</a>, user_function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia">GstRTSPSessionMedia</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMediaClass" title="struct GstRTSPSessionMediaClass">GstRTSPSessionMediaClass</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool-struct" title="struct GstRTSPSessionPool">GstRTSPSessionPool</a>, struct in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool-session-removed" title="The “session-removed” signal">GstRTSPSessionPool::session-removed</a>, object signal in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool--max-sessions" title="The “max-sessions” property">GstRTSPSessionPool:max-sessions</a>, object property in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolClass" title="struct GstRTSPSessionPoolClass">GstRTSPSessionPoolClass</a>, struct in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolFilterFunc" title="GstRTSPSessionPoolFilterFunc ()">GstRTSPSessionPoolFilterFunc</a>, user_function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolFunc" title="GstRTSPSessionPoolFunc ()">GstRTSPSessionPoolFunc</a>, user_function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#GstRTSPStream-struct" title="struct GstRTSPStream">GstRTSPStream</a>, struct in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#GstRTSPStreamClass" title="GstRTSPStreamClass">GstRTSPStreamClass</a>, struct in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport">GstRTSPStreamTransport</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransportClass" title="struct GstRTSPStreamTransportClass">GstRTSPStreamTransportClass</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#GstRTSPStreamTransportFilterFunc" title="GstRTSPStreamTransportFilterFunc ()">GstRTSPStreamTransportFilterFunc</a>, user_function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode">GstRTSPSuspendMode</a>, enum in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread">GstRTSPThread</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool">GstRTSPThreadPool</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPoolClass" title="struct GstRTSPThreadPoolClass">GstRTSPThreadPoolClass</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType">GstRTSPThreadType</a>, enum in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken">GstRTSPToken</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-copy" title="gst_rtsp_address_copy ()">gst_rtsp_address_copy</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free" title="gst_rtsp_address_free ()">gst_rtsp_address_free</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-acquire-address" title="gst_rtsp_address_pool_acquire_address ()">gst_rtsp_address_pool_acquire_address</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range" title="gst_rtsp_address_pool_add_range ()">gst_rtsp_address_pool_add_range</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV4:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV4">GST_RTSP_ADDRESS_POOL_ANY_IPV4</a>, macro in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV6:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV6">GST_RTSP_ADDRESS_POOL_ANY_IPV6</a>, macro in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-clear" title="gst_rtsp_address_pool_clear ()">gst_rtsp_address_pool_clear</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-dump" title="gst_rtsp_address_pool_dump ()">gst_rtsp_address_pool_dump</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-has-unicast-addresses" title="gst_rtsp_address_pool_has_unicast_addresses ()">gst_rtsp_address_pool_has_unicast_addresses</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-new" title="gst_rtsp_address_pool_new ()">gst_rtsp_address_pool_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-reserve-address" title="gst_rtsp_address_pool_reserve_address ()">gst_rtsp_address_pool_reserve_address</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-add-basic" title="gst_rtsp_auth_add_basic ()">gst_rtsp_auth_add_basic</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-check" title="gst_rtsp_auth_check ()">gst_rtsp_auth_check</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-CONNECT:CAPS" title="GST_RTSP_AUTH_CHECK_CONNECT">GST_RTSP_AUTH_CHECK_CONNECT</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-ACCESS:CAPS" title="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS">GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-CONSTRUCT:CAPS" title="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT">GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-TRANSPORT-CLIENT-SETTINGS:CAPS" title="GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS">GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-URL:CAPS" title="GST_RTSP_AUTH_CHECK_URL">GST_RTSP_AUTH_CHECK_URL</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-get-default-token" title="gst_rtsp_auth_get_default_token ()">gst_rtsp_auth_get_default_token</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-get-tls-certificate" title="gst_rtsp_auth_get_tls_certificate ()">gst_rtsp_auth_get_tls_certificate</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-make-basic" title="gst_rtsp_auth_make_basic ()">gst_rtsp_auth_make_basic</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-new" title="gst_rtsp_auth_new ()">gst_rtsp_auth_new</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-remove-basic" title="gst_rtsp_auth_remove_basic ()">gst_rtsp_auth_remove_basic</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-default-token" title="gst_rtsp_auth_set_default_token ()">gst_rtsp_auth_set_default_token</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate" title="gst_rtsp_auth_set_tls_certificate ()">gst_rtsp_auth_set_tls_certificate</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-attach" title="gst_rtsp_client_attach ()">gst_rtsp_client_attach</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-close" title="gst_rtsp_client_close ()">gst_rtsp_client_close</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-auth" title="gst_rtsp_client_get_auth ()">gst_rtsp_client_get_auth</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-connection" title="gst_rtsp_client_get_connection ()">gst_rtsp_client_get_connection</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-mount-points" title="gst_rtsp_client_get_mount_points ()">gst_rtsp_client_get_mount_points</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-session-pool" title="gst_rtsp_client_get_session_pool ()">gst_rtsp_client_get_session_pool</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-thread-pool" title="gst_rtsp_client_get_thread_pool ()">gst_rtsp_client_get_thread_pool</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-handle-message" title="gst_rtsp_client_handle_message ()">gst_rtsp_client_handle_message</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-new" title="gst_rtsp_client_new ()">gst_rtsp_client_new</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-send-message" title="gst_rtsp_client_send_message ()">gst_rtsp_client_send_message</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()">gst_rtsp_client_session_filter</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-auth" title="gst_rtsp_client_set_auth ()">gst_rtsp_client_set_auth</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-connection" title="gst_rtsp_client_set_connection ()">gst_rtsp_client_set_connection</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-mount-points" title="gst_rtsp_client_set_mount_points ()">gst_rtsp_client_set_mount_points</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-send-func" title="gst_rtsp_client_set_send_func ()">gst_rtsp_client_set_send_func</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-session-pool" title="gst_rtsp_client_set_session_pool ()">gst_rtsp_client_set_session_pool</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-thread-pool" title="gst_rtsp_client_set_thread_pool ()">gst_rtsp_client_set_thread_pool</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-get-current" title="gst_rtsp_context_get_current ()">gst_rtsp_context_get_current</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPContext.html" title="GstRTSPContext">GstRTSPContext</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-pop-current" title="gst_rtsp_context_pop_current ()">gst_rtsp_context_pop_current</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPContext.html" title="GstRTSPContext">GstRTSPContext</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-push-current" title="gst_rtsp_context_push_current ()">gst_rtsp_context_push_current</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPContext.html" title="GstRTSPContext">GstRTSPContext</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-collect-streams" title="gst_rtsp_media_collect_streams ()">gst_rtsp_media_collect_streams</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-create-stream" title="gst_rtsp_media_create_stream ()">gst_rtsp_media_create_stream</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-add-role" title="gst_rtsp_media_factory_add_role ()">gst_rtsp_media_factory_add_role</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-construct" title="gst_rtsp_media_factory_construct ()">gst_rtsp_media_factory_construct</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-create-element" title="gst_rtsp_media_factory_create_element ()">gst_rtsp_media_factory_create_element</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-address-pool" title="gst_rtsp_media_factory_get_address_pool ()">gst_rtsp_media_factory_get_address_pool</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-buffer-size" title="gst_rtsp_media_factory_get_buffer_size ()">gst_rtsp_media_factory_get_buffer_size</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-launch" title="gst_rtsp_media_factory_get_launch ()">gst_rtsp_media_factory_get_launch</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-permissions" title="gst_rtsp_media_factory_get_permissions ()">gst_rtsp_media_factory_get_permissions</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-profiles" title="gst_rtsp_media_factory_get_profiles ()">gst_rtsp_media_factory_get_profiles</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-protocols" title="gst_rtsp_media_factory_get_protocols ()">gst_rtsp_media_factory_get_protocols</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-suspend-mode" title="gst_rtsp_media_factory_get_suspend_mode ()">gst_rtsp_media_factory_get_suspend_mode</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-eos-shutdown" title="gst_rtsp_media_factory_is_eos_shutdown ()">gst_rtsp_media_factory_is_eos_shutdown</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-shared" title="gst_rtsp_media_factory_is_shared ()">gst_rtsp_media_factory_is_shared</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-new" title="gst_rtsp_media_factory_new ()">gst_rtsp_media_factory_new</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-address-pool" title="gst_rtsp_media_factory_set_address_pool ()">gst_rtsp_media_factory_set_address_pool</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-buffer-size" title="gst_rtsp_media_factory_set_buffer_size ()">gst_rtsp_media_factory_set_buffer_size</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-eos-shutdown" title="gst_rtsp_media_factory_set_eos_shutdown ()">gst_rtsp_media_factory_set_eos_shutdown</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-launch" title="gst_rtsp_media_factory_set_launch ()">gst_rtsp_media_factory_set_launch</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-permissions" title="gst_rtsp_media_factory_set_permissions ()">gst_rtsp_media_factory_set_permissions</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-profiles" title="gst_rtsp_media_factory_set_profiles ()">gst_rtsp_media_factory_set_profiles</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-protocols" title="gst_rtsp_media_factory_set_protocols ()">gst_rtsp_media_factory_set_protocols</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-shared" title="gst_rtsp_media_factory_set_shared ()">gst_rtsp_media_factory_set_shared</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-suspend-mode" title="gst_rtsp_media_factory_set_suspend_mode ()">gst_rtsp_media_factory_set_suspend_mode</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-get-uri" title="gst_rtsp_media_factory_uri_get_uri ()">gst_rtsp_media_factory_uri_get_uri</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-new" title="gst_rtsp_media_factory_uri_new ()">gst_rtsp_media_factory_uri_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-set-uri" title="gst_rtsp_media_factory_uri_set_uri ()">gst_rtsp_media_factory_uri_set_uri</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-find-stream" title="gst_rtsp_media_find_stream ()">gst_rtsp_media_find_stream</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-address-pool" title="gst_rtsp_media_get_address_pool ()">gst_rtsp_media_get_address_pool</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-base-time" title="gst_rtsp_media_get_base_time ()">gst_rtsp_media_get_base_time</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-buffer-size" title="gst_rtsp_media_get_buffer_size ()">gst_rtsp_media_get_buffer_size</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-clock" title="gst_rtsp_media_get_clock ()">gst_rtsp_media_get_clock</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-element" title="gst_rtsp_media_get_element ()">gst_rtsp_media_get_element</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-permissions" title="gst_rtsp_media_get_permissions ()">gst_rtsp_media_get_permissions</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-profiles" title="gst_rtsp_media_get_profiles ()">gst_rtsp_media_get_profiles</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-protocols" title="gst_rtsp_media_get_protocols ()">gst_rtsp_media_get_protocols</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-range-string" title="gst_rtsp_media_get_range_string ()">gst_rtsp_media_get_range_string</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-status" title="gst_rtsp_media_get_status ()">gst_rtsp_media_get_status</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-stream" title="gst_rtsp_media_get_stream ()">gst_rtsp_media_get_stream</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-suspend-mode" title="gst_rtsp_media_get_suspend_mode ()">gst_rtsp_media_get_suspend_mode</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-time-provider" title="gst_rtsp_media_get_time_provider ()">gst_rtsp_media_get_time_provider</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-eos-shutdown" title="gst_rtsp_media_is_eos_shutdown ()">gst_rtsp_media_is_eos_shutdown</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-reusable" title="gst_rtsp_media_is_reusable ()">gst_rtsp_media_is_reusable</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-shared" title="gst_rtsp_media_is_shared ()">gst_rtsp_media_is_shared</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-time-provider" title="gst_rtsp_media_is_time_provider ()">gst_rtsp_media_is_time_provider</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-new" title="gst_rtsp_media_new ()">gst_rtsp_media_new</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-n-streams" title="gst_rtsp_media_n_streams ()">gst_rtsp_media_n_streams</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()">gst_rtsp_media_prepare</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-seek" title="gst_rtsp_media_seek ()">gst_rtsp_media_seek</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-setup-sdp" title="gst_rtsp_media_setup_sdp ()">gst_rtsp_media_setup_sdp</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-address-pool" title="gst_rtsp_media_set_address_pool ()">gst_rtsp_media_set_address_pool</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-buffer-size" title="gst_rtsp_media_set_buffer_size ()">gst_rtsp_media_set_buffer_size</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-eos-shutdown" title="gst_rtsp_media_set_eos_shutdown ()">gst_rtsp_media_set_eos_shutdown</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-permissions" title="gst_rtsp_media_set_permissions ()">gst_rtsp_media_set_permissions</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-pipeline-state" title="gst_rtsp_media_set_pipeline_state ()">gst_rtsp_media_set_pipeline_state</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-profiles" title="gst_rtsp_media_set_profiles ()">gst_rtsp_media_set_profiles</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-protocols" title="gst_rtsp_media_set_protocols ()">gst_rtsp_media_set_protocols</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-reusable" title="gst_rtsp_media_set_reusable ()">gst_rtsp_media_set_reusable</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-shared" title="gst_rtsp_media_set_shared ()">gst_rtsp_media_set_shared</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-state" title="gst_rtsp_media_set_state ()">gst_rtsp_media_set_state</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-suspend-mode" title="gst_rtsp_media_set_suspend_mode ()">gst_rtsp_media_set_suspend_mode</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-suspend" title="gst_rtsp_media_suspend ()">gst_rtsp_media_suspend</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-take-pipeline" title="gst_rtsp_media_take_pipeline ()">gst_rtsp_media_take_pipeline</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unprepare" title="gst_rtsp_media_unprepare ()">gst_rtsp_media_unprepare</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unsuspend" title="gst_rtsp_media_unsuspend ()">gst_rtsp_media_unsuspend</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-use-time-provider" title="gst_rtsp_media_use_time_provider ()">gst_rtsp_media_use_time_provider</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-add-factory" title="gst_rtsp_mount_points_add_factory ()">gst_rtsp_mount_points_add_factory</a>, function in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-make-path" title="gst_rtsp_mount_points_make_path ()">gst_rtsp_mount_points_make_path</a>, function in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-match" title="gst_rtsp_mount_points_match ()">gst_rtsp_mount_points_match</a>, function in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-new" title="gst_rtsp_mount_points_new ()">gst_rtsp_mount_points_new</a>, function in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-remove-factory" title="gst_rtsp_mount_points_remove_factory ()">gst_rtsp_mount_points_remove_factory</a>, function in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-get" title="gst_rtsp_params_get ()">gst_rtsp_params_get</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPParams.html" title="GstRTSPParams">GstRTSPParams</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-set" title="gst_rtsp_params_set ()">gst_rtsp_params_set</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPParams.html" title="GstRTSPParams">GstRTSPParams</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role" title="gst_rtsp_permissions_add_role ()">gst_rtsp_permissions_add_role</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role-valist" title="gst_rtsp_permissions_add_role_valist ()">gst_rtsp_permissions_add_role_valist</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-get-role" title="gst_rtsp_permissions_get_role ()">gst_rtsp_permissions_get_role</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-is-allowed" title="gst_rtsp_permissions_is_allowed ()">gst_rtsp_permissions_is_allowed</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-new" title="gst_rtsp_permissions_new ()">gst_rtsp_permissions_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-ref" title="gst_rtsp_permissions_ref ()">gst_rtsp_permissions_ref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-remove-role" title="gst_rtsp_permissions_remove_role ()">gst_rtsp_permissions_remove_role</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-unref" title="gst_rtsp_permissions_unref ()">gst_rtsp_permissions_unref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-ACCESS:CAPS" title="GST_RTSP_PERM_MEDIA_FACTORY_ACCESS">GST_RTSP_PERM_MEDIA_FACTORY_ACCESS</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-CONSTRUCT:CAPS" title="GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT">GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSdp.html#gst-rtsp-sdp-from-media" title="gst_rtsp_sdp_from_media ()">gst_rtsp_sdp_from_media</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSdp.html" title="GstRTSPSdp">GstRTSPSdp</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-attach" title="gst_rtsp_server_attach ()">gst_rtsp_server_attach</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-client-filter" title="gst_rtsp_server_client_filter ()">gst_rtsp_server_client_filter</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-socket" title="gst_rtsp_server_create_socket ()">gst_rtsp_server_create_socket</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-source" title="gst_rtsp_server_create_source ()">gst_rtsp_server_create_source</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-address" title="gst_rtsp_server_get_address ()">gst_rtsp_server_get_address</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-auth" title="gst_rtsp_server_get_auth ()">gst_rtsp_server_get_auth</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-backlog" title="gst_rtsp_server_get_backlog ()">gst_rtsp_server_get_backlog</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-bound-port" title="gst_rtsp_server_get_bound_port ()">gst_rtsp_server_get_bound_port</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-mount-points" title="gst_rtsp_server_get_mount_points ()">gst_rtsp_server_get_mount_points</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-service" title="gst_rtsp_server_get_service ()">gst_rtsp_server_get_service</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-session-pool" title="gst_rtsp_server_get_session_pool ()">gst_rtsp_server_get_session_pool</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-thread-pool" title="gst_rtsp_server_get_thread_pool ()">gst_rtsp_server_get_thread_pool</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-io-func" title="gst_rtsp_server_io_func ()">gst_rtsp_server_io_func</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-new" title="gst_rtsp_server_new ()">gst_rtsp_server_new</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-address" title="gst_rtsp_server_set_address ()">gst_rtsp_server_set_address</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-auth" title="gst_rtsp_server_set_auth ()">gst_rtsp_server_set_auth</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-backlog" title="gst_rtsp_server_set_backlog ()">gst_rtsp_server_set_backlog</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-mount-points" title="gst_rtsp_server_set_mount_points ()">gst_rtsp_server_set_mount_points</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-service" title="gst_rtsp_server_set_service ()">gst_rtsp_server_set_service</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-session-pool" title="gst_rtsp_server_set_session_pool ()">gst_rtsp_server_set_session_pool</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-thread-pool" title="gst_rtsp_server_set_thread_pool ()">gst_rtsp_server_set_thread_pool</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPServer.html#gst-rtsp-server-transfer-connection" title="gst_rtsp_server_transfer_connection ()">gst_rtsp_server_transfer_connection</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-allow-expire" title="gst_rtsp_session_allow_expire ()">gst_rtsp_session_allow_expire</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()">gst_rtsp_session_filter</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-header" title="gst_rtsp_session_get_header ()">gst_rtsp_session_get_header</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-media" title="gst_rtsp_session_get_media ()">gst_rtsp_session_get_media</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-sessionid" title="gst_rtsp_session_get_sessionid ()">gst_rtsp_session_get_sessionid</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-timeout" title="gst_rtsp_session_get_timeout ()">gst_rtsp_session_get_timeout</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-is-expired" title="gst_rtsp_session_is_expired ()">gst_rtsp_session_is_expired</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-manage-media" title="gst_rtsp_session_manage_media ()">gst_rtsp_session_manage_media</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-alloc-channels" title="gst_rtsp_session_media_alloc_channels ()">gst_rtsp_session_media_alloc_channels</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-base-time" title="gst_rtsp_session_media_get_base_time ()">gst_rtsp_session_media_get_base_time</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-media" title="gst_rtsp_session_media_get_media ()">gst_rtsp_session_media_get_media</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtpinfo" title="gst_rtsp_session_media_get_rtpinfo ()">gst_rtsp_session_media_get_rtpinfo</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtsp-state" title="gst_rtsp_session_media_get_rtsp_state ()">gst_rtsp_session_media_get_rtsp_state</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-transport" title="gst_rtsp_session_media_get_transport ()">gst_rtsp_session_media_get_transport</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-matches" title="gst_rtsp_session_media_matches ()">gst_rtsp_session_media_matches</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-new" title="gst_rtsp_session_media_new ()">gst_rtsp_session_media_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-rtsp-state" title="gst_rtsp_session_media_set_rtsp_state ()">gst_rtsp_session_media_set_rtsp_state</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-state" title="gst_rtsp_session_media_set_state ()">gst_rtsp_session_media_set_state</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-transport" title="gst_rtsp_session_media_set_transport ()">gst_rtsp_session_media_set_transport</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-new" title="gst_rtsp_session_new ()">gst_rtsp_session_new</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-next-timeout" title="gst_rtsp_session_next_timeout ()">gst_rtsp_session_next_timeout</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-cleanup" title="gst_rtsp_session_pool_cleanup ()">gst_rtsp_session_pool_cleanup</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create" title="gst_rtsp_session_pool_create ()">gst_rtsp_session_pool_create</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create-watch" title="gst_rtsp_session_pool_create_watch ()">gst_rtsp_session_pool_create_watch</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()">gst_rtsp_session_pool_filter</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-find" title="gst_rtsp_session_pool_find ()">gst_rtsp_session_pool_find</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-max-sessions" title="gst_rtsp_session_pool_get_max_sessions ()">gst_rtsp_session_pool_get_max_sessions</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-n-sessions" title="gst_rtsp_session_pool_get_n_sessions ()">gst_rtsp_session_pool_get_n_sessions</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-new" title="gst_rtsp_session_pool_new ()">gst_rtsp_session_pool_new</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-remove" title="gst_rtsp_session_pool_remove ()">gst_rtsp_session_pool_remove</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-set-max-sessions" title="gst_rtsp_session_pool_set_max_sessions ()">gst_rtsp_session_pool_set_max_sessions</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-prevent-expire" title="gst_rtsp_session_prevent_expire ()">gst_rtsp_session_prevent_expire</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-release-media" title="gst_rtsp_session_release_media ()">gst_rtsp_session_release_media</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-set-timeout" title="gst_rtsp_session_set_timeout ()">gst_rtsp_session_set_timeout</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPSession.html#gst-rtsp-session-touch" title="gst_rtsp_session_touch ()">gst_rtsp_session_touch</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-add-transport" title="gst_rtsp_stream_add_transport ()">gst_rtsp_stream_add_transport</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-address-pool" title="gst_rtsp_stream_get_address_pool ()">gst_rtsp_stream_get_address_pool</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-caps" title="gst_rtsp_stream_get_caps ()">gst_rtsp_stream_get_caps</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-control" title="gst_rtsp_stream_get_control ()">gst_rtsp_stream_get_control</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-dscp-qos" title="gst_rtsp_stream_get_dscp_qos ()">gst_rtsp_stream_get_dscp_qos</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-index" title="gst_rtsp_stream_get_index ()">gst_rtsp_stream_get_index</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-mtu" title="gst_rtsp_stream_get_mtu ()">gst_rtsp_stream_get_mtu</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-multicast-address" title="gst_rtsp_stream_get_multicast_address ()">gst_rtsp_stream_get_multicast_address</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-profiles" title="gst_rtsp_stream_get_profiles ()">gst_rtsp_stream_get_profiles</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-protocols" title="gst_rtsp_stream_get_protocols ()">gst_rtsp_stream_get_protocols</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-pt" title="gst_rtsp_stream_get_pt ()">gst_rtsp_stream_get_pt</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtcp-socket" title="gst_rtsp_stream_get_rtcp_socket ()">gst_rtsp_stream_get_rtcp_socket</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtpinfo" title="gst_rtsp_stream_get_rtpinfo ()">gst_rtsp_stream_get_rtpinfo</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtpsession" title="gst_rtsp_stream_get_rtpsession ()">gst_rtsp_stream_get_rtpsession</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtp-socket" title="gst_rtsp_stream_get_rtp_socket ()">gst_rtsp_stream_get_rtp_socket</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-server-port" title="gst_rtsp_stream_get_server_port ()">gst_rtsp_stream_get_server_port</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-srcpad" title="gst_rtsp_stream_get_srcpad ()">gst_rtsp_stream_get_srcpad</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-ssrc" title="gst_rtsp_stream_get_ssrc ()">gst_rtsp_stream_get_ssrc</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-has-control" title="gst_rtsp_stream_has_control ()">gst_rtsp_stream_has_control</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-is-blocking" title="gst_rtsp_stream_is_blocking ()">gst_rtsp_stream_is_blocking</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-is-transport-supported" title="gst_rtsp_stream_is_transport_supported ()">gst_rtsp_stream_is_transport_supported</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-join-bin" title="gst_rtsp_stream_join_bin ()">gst_rtsp_stream_join_bin</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-leave-bin" title="gst_rtsp_stream_leave_bin ()">gst_rtsp_stream_leave_bin</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-new" title="gst_rtsp_stream_new ()">gst_rtsp_stream_new</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-recv-rtcp" title="gst_rtsp_stream_recv_rtcp ()">gst_rtsp_stream_recv_rtcp</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-recv-rtp" title="gst_rtsp_stream_recv_rtp ()">gst_rtsp_stream_recv_rtp</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-remove-transport" title="gst_rtsp_stream_remove_transport ()">gst_rtsp_stream_remove_transport</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-reserve-address" title="gst_rtsp_stream_reserve_address ()">gst_rtsp_stream_reserve_address</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-address-pool" title="gst_rtsp_stream_set_address_pool ()">gst_rtsp_stream_set_address_pool</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-blocked" title="gst_rtsp_stream_set_blocked ()">gst_rtsp_stream_set_blocked</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-control" title="gst_rtsp_stream_set_control ()">gst_rtsp_stream_set_control</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-dscp-qos" title="gst_rtsp_stream_set_dscp_qos ()">gst_rtsp_stream_set_dscp_qos</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-mtu" title="gst_rtsp_stream_set_mtu ()">gst_rtsp_stream_set_mtu</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-profiles" title="gst_rtsp_stream_set_profiles ()">gst_rtsp_stream_set_profiles</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-protocols" title="gst_rtsp_stream_set_protocols ()">gst_rtsp_stream_set_protocols</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-transport-filter" title="gst_rtsp_stream_transport_filter ()">gst_rtsp_stream_transport_filter</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-rtpinfo" title="gst_rtsp_stream_transport_get_rtpinfo ()">gst_rtsp_stream_transport_get_rtpinfo</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-stream" title="gst_rtsp_stream_transport_get_stream ()">gst_rtsp_stream_transport_get_stream</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-transport" title="gst_rtsp_stream_transport_get_transport ()">gst_rtsp_stream_transport_get_transport</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-url" title="gst_rtsp_stream_transport_get_url ()">gst_rtsp_stream_transport_get_url</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-is-timed-out" title="gst_rtsp_stream_transport_is_timed_out ()">gst_rtsp_stream_transport_is_timed_out</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-keep-alive" title="gst_rtsp_stream_transport_keep_alive ()">gst_rtsp_stream_transport_keep_alive</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-new" title="gst_rtsp_stream_transport_new ()">gst_rtsp_stream_transport_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtcp" title="gst_rtsp_stream_transport_send_rtcp ()">gst_rtsp_stream_transport_send_rtcp</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtp" title="gst_rtsp_stream_transport_send_rtp ()">gst_rtsp_stream_transport_send_rtp</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-active" title="gst_rtsp_stream_transport_set_active ()">gst_rtsp_stream_transport_set_active</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-callbacks" title="gst_rtsp_stream_transport_set_callbacks ()">gst_rtsp_stream_transport_set_callbacks</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-keepalive" title="gst_rtsp_stream_transport_set_keepalive ()">gst_rtsp_stream_transport_set_keepalive</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-timed-out" title="gst_rtsp_stream_transport_set_timed_out ()">gst_rtsp_stream_transport_set_timed_out</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-transport" title="gst_rtsp_stream_transport_set_transport ()">gst_rtsp_stream_transport_set_transport</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-url" title="gst_rtsp_stream_transport_set_url ()">gst_rtsp_stream_transport_set_url</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-update-crypto" title="gst_rtsp_stream_update_crypto ()">gst_rtsp_stream_update_crypto</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-new" title="gst_rtsp_thread_new ()">gst_rtsp_thread_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-cleanup" title="gst_rtsp_thread_pool_cleanup ()">gst_rtsp_thread_pool_cleanup</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-max-threads" title="gst_rtsp_thread_pool_get_max_threads ()">gst_rtsp_thread_pool_get_max_threads</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-thread" title="gst_rtsp_thread_pool_get_thread ()">gst_rtsp_thread_pool_get_thread</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-new" title="gst_rtsp_thread_pool_new ()">gst_rtsp_thread_pool_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-set-max-threads" title="gst_rtsp_thread_pool_set_max_threads ()">gst_rtsp_thread_pool_set_max_threads</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-ref" title="gst_rtsp_thread_ref ()">gst_rtsp_thread_ref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-reuse" title="gst_rtsp_thread_reuse ()">gst_rtsp_thread_reuse</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-stop" title="gst_rtsp_thread_stop ()">gst_rtsp_thread_stop</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-unref" title="gst_rtsp_thread_unref ()">gst_rtsp_thread_unref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-string" title="gst_rtsp_token_get_string ()">gst_rtsp_token_get_string</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-structure" title="gst_rtsp_token_get_structure ()">gst_rtsp_token_get_structure</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-is-allowed" title="gst_rtsp_token_is_allowed ()">gst_rtsp_token_is_allowed</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GST-RTSP-TOKEN-MEDIA-FACTORY-ROLE:CAPS" title="GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE">GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new" title="gst_rtsp_token_new ()">gst_rtsp_token_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-empty" title="gst_rtsp_token_new_empty ()">gst_rtsp_token_new_empty</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-valist" title="gst_rtsp_token_new_valist ()">gst_rtsp_token_new_valist</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-ref" title="gst_rtsp_token_ref ()">gst_rtsp_token_ref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="GstRTSPAuth.html#GST-RTSP-TOKEN-TRANSPORT-CLIENT-SETTINGS:CAPS" title="GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS">GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-unref" title="gst_rtsp_token_unref ()">gst_rtsp_token_unref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
+</dt>
+<dd></dd>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-writable-structure" title="gst_rtsp_token_writable_structure ()">gst_rtsp_token_writable_structure</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
+</dt>
+<dd></dd>
+<a name="idxS"></a><h3 class="title">S</h3>
+<dt>
+<a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo">GstSDPInfo</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPSdp.html" title="GstRTSPSdp">GstRTSPSdp</a>
+</dt>
+<dd></dd>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
\ No newline at end of file
--- /dev/null
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
+<html>
+<head>
+<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
+<title>GStreamer RTSP Server Reference Manual: </title>
+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
+<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
+<link rel="up" href="index.html" title="GStreamer RTSP Server Reference Manual">
+<link rel="prev" href="index.html" title="GStreamer RTSP Server Reference Manual">
+<link rel="next" href="GstRTSPServer.html" title="GstRTSPServer">
+<meta name="generator" content="GTK-Doc V1.21 (XML mode)">
+<link rel="stylesheet" href="style.css" type="text/css">
+</head>
+<body bgcolor="white" text="black" link="#0000FF" vlink="#840084" alink="#0000FF">
+<table class="navigation" id="top" width="100%" summary="Navigation header" cellpadding="2" cellspacing="5"><tr valign="middle">
+<td width="100%" align="left" class="shortcuts"></td>
+<td><a accesskey="h" href="index.html"><img src="home.png" width="16" height="16" border="0" alt="Home"></a></td>
+<td><img src="up-insensitive.png" width="16" height="16" border="0"></td>
+<td><a accesskey="p" href="index.html"><img src="left.png" width="16" height="16" border="0" alt="Prev"></a></td>
+<td><a accesskey="n" href="GstRTSPServer.html"><img src="right.png" width="16" height="16" border="0" alt="Next"></a></td>
+</tr></table>
+<div class="chapter">
+<div class="titlepage"></div>
+<div class="toc"><dl class="toc">
+<dt>
+<span class="refentrytitle"><a href="GstRTSPServer.html">GstRTSPServer</a></span><span class="refpurpose"> — The main server object</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPClient.html">GstRTSPClient</a></span><span class="refpurpose"> — A client connection state</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPContext.html">GstRTSPContext</a></span><span class="refpurpose"> — A client request context</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPMountPoints.html">GstRTSPMountPoints</a></span><span class="refpurpose"> — Map a path to media</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPMediaFactory.html">GstRTSPMediaFactory</a></span><span class="refpurpose"> — A factory for media pipelines</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPMediaFactoryURI.html">GstRTSPMediaFactoryURI</a></span><span class="refpurpose"> — A factory for URI sources</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPMedia.html">GstRTSPMedia</a></span><span class="refpurpose"> — The media pipeline</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPStream.html">GstRTSPStream</a></span><span class="refpurpose"> — A media stream</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPSessionPool.html">GstRTSPSessionPool</a></span><span class="refpurpose"> — An object for managing sessions</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPSession.html">GstRTSPSession</a></span><span class="refpurpose"> — An object to manage media</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPSessionMedia.html">GstRTSPSessionMedia</a></span><span class="refpurpose"> — Media managed in a session</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPStreamTransport.html">GstRTSPStreamTransport</a></span><span class="refpurpose"> — A media stream transport configuration</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPSdp.html">GstRTSPSdp</a></span><span class="refpurpose"> — Make SDP messages</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPAddressPool.html">GstRTSPAddressPool</a></span><span class="refpurpose"> — A pool of network addresses</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPThreadPool.html">GstRTSPThreadPool</a></span><span class="refpurpose"> — A pool of threads</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPAuth.html">GstRTSPAuth</a></span><span class="refpurpose"> — Authentication and authorization</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPToken.html">GstRTSPToken</a></span><span class="refpurpose"> — Roles and permissions for a client</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPPermissions.html">GstRTSPPermissions</a></span><span class="refpurpose"> — Roles and associated permissions</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPParams.html">GstRTSPParams</a></span><span class="refpurpose"> — Param get and set implementation</span>
+</dt>
+</dl></div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
\ No newline at end of file
--- /dev/null
+<?xml version="1.0" encoding="utf-8" standalone="no"?>
+<!DOCTYPE book PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "">
+<book xmlns="http://www.devhelp.net/book" title="GStreamer RTSP Server Reference Manual" link="index.html" author="" name="gst-rtsp-server-1.0" version="2" language="c">
+ <chapters>
+ <sub name="" link="ch01.html">
+ <sub name="GstRTSPServer" link="GstRTSPServer.html"/>
+ <sub name="GstRTSPClient" link="GstRTSPClient.html"/>
+ <sub name="GstRTSPContext" link="gst-rtsp-server-GstRTSPContext.html"/>
+ <sub name="GstRTSPMountPoints" link="GstRTSPMountPoints.html"/>
+ <sub name="GstRTSPMediaFactory" link="GstRTSPMediaFactory.html"/>
+ <sub name="GstRTSPMediaFactoryURI" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html"/>
+ <sub name="GstRTSPMedia" link="GstRTSPMedia.html"/>
+ <sub name="GstRTSPStream" link="GstRTSPStream.html"/>
+ <sub name="GstRTSPSessionPool" link="GstRTSPSessionPool.html"/>
+ <sub name="GstRTSPSession" link="GstRTSPSession.html"/>
+ <sub name="GstRTSPSessionMedia" link="gst-rtsp-server-GstRTSPSessionMedia.html"/>
+ <sub name="GstRTSPStreamTransport" link="gst-rtsp-server-GstRTSPStreamTransport.html"/>
+ <sub name="GstRTSPSdp" link="gst-rtsp-server-GstRTSPSdp.html"/>
+ <sub name="GstRTSPAddressPool" link="gst-rtsp-server-GstRTSPAddressPool.html"/>
+ <sub name="GstRTSPThreadPool" link="gst-rtsp-server-GstRTSPThreadPool.html"/>
+ <sub name="GstRTSPAuth" link="GstRTSPAuth.html"/>
+ <sub name="GstRTSPToken" link="gst-rtsp-server-GstRTSPToken.html"/>
+ <sub name="GstRTSPPermissions" link="gst-rtsp-server-GstRTSPPermissions.html"/>
+ <sub name="GstRTSPParams" link="gst-rtsp-server-GstRTSPParams.html"/>
+ </sub>
+ <sub name="Object Hierarchy" link="rtsp-server-hierarchy.html"/>
+ <sub name="API Index" link="api-index-full.html"/>
+ <sub name="Annotation Glossary" link="annotation-glossary.html"/>
+ </chapters>
+ <functions>
+ <keyword type="function" name="gst_rtsp_server_new ()" link="GstRTSPServer.html#gst-rtsp-server-new"/>
+ <keyword type="function" name="gst_rtsp_server_get_address ()" link="GstRTSPServer.html#gst-rtsp-server-get-address"/>
+ <keyword type="function" name="gst_rtsp_server_set_address ()" link="GstRTSPServer.html#gst-rtsp-server-set-address"/>
+ <keyword type="function" name="gst_rtsp_server_get_service ()" link="GstRTSPServer.html#gst-rtsp-server-get-service"/>
+ <keyword type="function" name="gst_rtsp_server_set_service ()" link="GstRTSPServer.html#gst-rtsp-server-set-service"/>
+ <keyword type="function" name="gst_rtsp_server_get_backlog ()" link="GstRTSPServer.html#gst-rtsp-server-get-backlog"/>
+ <keyword type="function" name="gst_rtsp_server_set_backlog ()" link="GstRTSPServer.html#gst-rtsp-server-set-backlog"/>
+ <keyword type="function" name="gst_rtsp_server_get_bound_port ()" link="GstRTSPServer.html#gst-rtsp-server-get-bound-port"/>
+ <keyword type="function" name="gst_rtsp_server_get_mount_points ()" link="GstRTSPServer.html#gst-rtsp-server-get-mount-points"/>
+ <keyword type="function" name="gst_rtsp_server_set_mount_points ()" link="GstRTSPServer.html#gst-rtsp-server-set-mount-points"/>
+ <keyword type="function" name="gst_rtsp_server_get_session_pool ()" link="GstRTSPServer.html#gst-rtsp-server-get-session-pool"/>
+ <keyword type="function" name="gst_rtsp_server_set_session_pool ()" link="GstRTSPServer.html#gst-rtsp-server-set-session-pool"/>
+ <keyword type="function" name="gst_rtsp_server_get_thread_pool ()" link="GstRTSPServer.html#gst-rtsp-server-get-thread-pool"/>
+ <keyword type="function" name="gst_rtsp_server_set_thread_pool ()" link="GstRTSPServer.html#gst-rtsp-server-set-thread-pool"/>
+ <keyword type="function" name="gst_rtsp_server_get_auth ()" link="GstRTSPServer.html#gst-rtsp-server-get-auth"/>
+ <keyword type="function" name="gst_rtsp_server_set_auth ()" link="GstRTSPServer.html#gst-rtsp-server-set-auth"/>
+ <keyword type="function" name="gst_rtsp_server_transfer_connection ()" link="GstRTSPServer.html#gst-rtsp-server-transfer-connection"/>
+ <keyword type="function" name="gst_rtsp_server_io_func ()" link="GstRTSPServer.html#gst-rtsp-server-io-func"/>
+ <keyword type="function" name="gst_rtsp_server_create_socket ()" link="GstRTSPServer.html#gst-rtsp-server-create-socket"/>
+ <keyword type="function" name="gst_rtsp_server_create_source ()" link="GstRTSPServer.html#gst-rtsp-server-create-source"/>
+ <keyword type="function" name="gst_rtsp_server_attach ()" link="GstRTSPServer.html#gst-rtsp-server-attach"/>
+ <keyword type="function" name="GstRTSPServerClientFilterFunc ()" link="GstRTSPServer.html#GstRTSPServerClientFilterFunc"/>
+ <keyword type="function" name="gst_rtsp_server_client_filter ()" link="GstRTSPServer.html#gst-rtsp-server-client-filter"/>
+ <keyword type="struct" name="struct GstRTSPServer" link="GstRTSPServer.html#GstRTSPServer-struct"/>
+ <keyword type="struct" name="struct GstRTSPServerClass" link="GstRTSPServer.html#GstRTSPServerClass"/>
+ <keyword type="property" name="The “address” property" link="GstRTSPServer.html#GstRTSPServer--address"/>
+ <keyword type="property" name="The “backlog” property" link="GstRTSPServer.html#GstRTSPServer--backlog"/>
+ <keyword type="property" name="The “bound-port” property" link="GstRTSPServer.html#GstRTSPServer--bound-port"/>
+ <keyword type="property" name="The “mount-points” property" link="GstRTSPServer.html#GstRTSPServer--mount-points"/>
+ <keyword type="property" name="The “service” property" link="GstRTSPServer.html#GstRTSPServer--service"/>
+ <keyword type="property" name="The “session-pool” property" link="GstRTSPServer.html#GstRTSPServer--session-pool"/>
+ <keyword type="signal" name="The “client-connected” signal" link="GstRTSPServer.html#GstRTSPServer-client-connected"/>
+ <keyword type="function" name="gst_rtsp_client_new ()" link="GstRTSPClient.html#gst-rtsp-client-new"/>
+ <keyword type="function" name="gst_rtsp_client_close ()" link="GstRTSPClient.html#gst-rtsp-client-close" since="1.4"/>
+ <keyword type="function" name="gst_rtsp_client_get_session_pool ()" link="GstRTSPClient.html#gst-rtsp-client-get-session-pool"/>
+ <keyword type="function" name="gst_rtsp_client_set_session_pool ()" link="GstRTSPClient.html#gst-rtsp-client-set-session-pool"/>
+ <keyword type="function" name="gst_rtsp_client_get_mount_points ()" link="GstRTSPClient.html#gst-rtsp-client-get-mount-points"/>
+ <keyword type="function" name="gst_rtsp_client_set_mount_points ()" link="GstRTSPClient.html#gst-rtsp-client-set-mount-points"/>
+ <keyword type="function" name="gst_rtsp_client_get_auth ()" link="GstRTSPClient.html#gst-rtsp-client-get-auth"/>
+ <keyword type="function" name="gst_rtsp_client_set_auth ()" link="GstRTSPClient.html#gst-rtsp-client-set-auth"/>
+ <keyword type="function" name="gst_rtsp_client_get_thread_pool ()" link="GstRTSPClient.html#gst-rtsp-client-get-thread-pool"/>
+ <keyword type="function" name="gst_rtsp_client_set_thread_pool ()" link="GstRTSPClient.html#gst-rtsp-client-set-thread-pool"/>
+ <keyword type="function" name="gst_rtsp_client_get_connection ()" link="GstRTSPClient.html#gst-rtsp-client-get-connection"/>
+ <keyword type="function" name="gst_rtsp_client_set_connection ()" link="GstRTSPClient.html#gst-rtsp-client-set-connection"/>
+ <keyword type="function" name="gst_rtsp_client_attach ()" link="GstRTSPClient.html#gst-rtsp-client-attach"/>
+ <keyword type="function" name="GstRTSPClientSendFunc ()" link="GstRTSPClient.html#GstRTSPClientSendFunc"/>
+ <keyword type="function" name="gst_rtsp_client_set_send_func ()" link="GstRTSPClient.html#gst-rtsp-client-set-send-func"/>
+ <keyword type="function" name="gst_rtsp_client_handle_message ()" link="GstRTSPClient.html#gst-rtsp-client-handle-message"/>
+ <keyword type="function" name="gst_rtsp_client_send_message ()" link="GstRTSPClient.html#gst-rtsp-client-send-message"/>
+ <keyword type="function" name="GstRTSPClientSessionFilterFunc ()" link="GstRTSPClient.html#GstRTSPClientSessionFilterFunc"/>
+ <keyword type="function" name="gst_rtsp_client_session_filter ()" link="GstRTSPClient.html#gst-rtsp-client-session-filter"/>
+ <keyword type="struct" name="struct GstRTSPClient" link="GstRTSPClient.html#GstRTSPClient-struct"/>
+ <keyword type="struct" name="struct GstRTSPClientClass" link="GstRTSPClient.html#GstRTSPClientClass"/>
+ <keyword type="property" name="The “drop-backlog” property" link="GstRTSPClient.html#GstRTSPClient--drop-backlog"/>
+ <keyword type="property" name="The “mount-points” property" link="GstRTSPClient.html#GstRTSPClient--mount-points"/>
+ <keyword type="property" name="The “session-pool” property" link="GstRTSPClient.html#GstRTSPClient--session-pool"/>
+ <keyword type="signal" name="The “closed” signal" link="GstRTSPClient.html#GstRTSPClient-closed"/>
+ <keyword type="signal" name="The “describe-request” signal" link="GstRTSPClient.html#GstRTSPClient-describe-request"/>
+ <keyword type="signal" name="The “get-parameter-request” signal" link="GstRTSPClient.html#GstRTSPClient-get-parameter-request"/>
+ <keyword type="signal" name="The “handle-response” signal" link="GstRTSPClient.html#GstRTSPClient-handle-response"/>
+ <keyword type="signal" name="The “new-session” signal" link="GstRTSPClient.html#GstRTSPClient-new-session"/>
+ <keyword type="signal" name="The “options-request” signal" link="GstRTSPClient.html#GstRTSPClient-options-request"/>
+ <keyword type="signal" name="The “pause-request” signal" link="GstRTSPClient.html#GstRTSPClient-pause-request"/>
+ <keyword type="signal" name="The “play-request” signal" link="GstRTSPClient.html#GstRTSPClient-play-request"/>
+ <keyword type="signal" name="The “send-message” signal" link="GstRTSPClient.html#GstRTSPClient-send-message"/>
+ <keyword type="signal" name="The “set-parameter-request” signal" link="GstRTSPClient.html#GstRTSPClient-set-parameter-request"/>
+ <keyword type="signal" name="The “setup-request” signal" link="GstRTSPClient.html#GstRTSPClient-setup-request"/>
+ <keyword type="signal" name="The “teardown-request” signal" link="GstRTSPClient.html#GstRTSPClient-teardown-request"/>
+ <keyword type="function" name="gst_rtsp_context_get_current ()" link="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-get-current"/>
+ <keyword type="function" name="gst_rtsp_context_push_current ()" link="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-push-current"/>
+ <keyword type="function" name="gst_rtsp_context_pop_current ()" link="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-pop-current"/>
+ <keyword type="struct" name="struct GstRTSPContext" link="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext"/>
+ <keyword type="function" name="gst_rtsp_mount_points_new ()" link="GstRTSPMountPoints.html#gst-rtsp-mount-points-new"/>
+ <keyword type="function" name="gst_rtsp_mount_points_add_factory ()" link="GstRTSPMountPoints.html#gst-rtsp-mount-points-add-factory"/>
+ <keyword type="function" name="gst_rtsp_mount_points_remove_factory ()" link="GstRTSPMountPoints.html#gst-rtsp-mount-points-remove-factory"/>
+ <keyword type="function" name="gst_rtsp_mount_points_match ()" link="GstRTSPMountPoints.html#gst-rtsp-mount-points-match"/>
+ <keyword type="function" name="gst_rtsp_mount_points_make_path ()" link="GstRTSPMountPoints.html#gst-rtsp-mount-points-make-path"/>
+ <keyword type="struct" name="struct GstRTSPMountPoints" link="GstRTSPMountPoints.html#GstRTSPMountPoints-struct"/>
+ <keyword type="struct" name="struct GstRTSPMountPointsClass" link="GstRTSPMountPoints.html#GstRTSPMountPointsClass"/>
+ <keyword type="function" name="gst_rtsp_media_factory_new ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-new"/>
+ <keyword type="function" name="gst_rtsp_media_factory_get_launch ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-launch"/>
+ <keyword type="function" name="gst_rtsp_media_factory_set_launch ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-launch"/>
+ <keyword type="function" name="gst_rtsp_media_factory_get_permissions ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-permissions"/>
+ <keyword type="function" name="gst_rtsp_media_factory_set_permissions ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-permissions"/>
+ <keyword type="function" name="gst_rtsp_media_factory_add_role ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-add-role"/>
+ <keyword type="function" name="gst_rtsp_media_factory_set_shared ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-shared"/>
+ <keyword type="function" name="gst_rtsp_media_factory_is_shared ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-shared"/>
+ <keyword type="function" name="gst_rtsp_media_factory_is_eos_shutdown ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-eos-shutdown"/>
+ <keyword type="function" name="gst_rtsp_media_factory_set_eos_shutdown ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-eos-shutdown"/>
+ <keyword type="function" name="gst_rtsp_media_factory_get_protocols ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-protocols"/>
+ <keyword type="function" name="gst_rtsp_media_factory_set_protocols ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-protocols"/>
+ <keyword type="function" name="gst_rtsp_media_factory_set_profiles ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-profiles"/>
+ <keyword type="function" name="gst_rtsp_media_factory_get_profiles ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-profiles"/>
+ <keyword type="function" name="gst_rtsp_media_factory_get_address_pool ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-address-pool"/>
+ <keyword type="function" name="gst_rtsp_media_factory_set_address_pool ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-address-pool"/>
+ <keyword type="function" name="gst_rtsp_media_factory_get_buffer_size ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-buffer-size"/>
+ <keyword type="function" name="gst_rtsp_media_factory_set_buffer_size ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-buffer-size"/>
+ <keyword type="function" name="gst_rtsp_media_factory_get_suspend_mode ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-suspend-mode"/>
+ <keyword type="function" name="gst_rtsp_media_factory_set_suspend_mode ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-suspend-mode"/>
+ <keyword type="function" name="gst_rtsp_media_factory_construct ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-construct"/>
+ <keyword type="function" name="gst_rtsp_media_factory_create_element ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-create-element"/>
+ <keyword type="struct" name="struct GstRTSPMediaFactory" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory-struct"/>
+ <keyword type="struct" name="struct GstRTSPMediaFactoryClass" link="GstRTSPMediaFactory.html#GstRTSPMediaFactoryClass"/>
+ <keyword type="property" name="The “buffer-size” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--buffer-size"/>
+ <keyword type="property" name="The “eos-shutdown” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--eos-shutdown"/>
+ <keyword type="property" name="The “launch” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--launch"/>
+ <keyword type="property" name="The “profiles” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--profiles"/>
+ <keyword type="property" name="The “protocols” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--protocols"/>
+ <keyword type="property" name="The “shared” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--shared"/>
+ <keyword type="property" name="The “suspend-mode” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--suspend-mode"/>
+ <keyword type="signal" name="The “media-configure” signal" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-configure"/>
+ <keyword type="signal" name="The “media-constructed” signal" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-constructed"/>
+ <keyword type="function" name="gst_rtsp_media_factory_uri_new ()" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-new"/>
+ <keyword type="function" name="gst_rtsp_media_factory_uri_set_uri ()" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-set-uri"/>
+ <keyword type="function" name="gst_rtsp_media_factory_uri_get_uri ()" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-get-uri"/>
+ <keyword type="struct" name="struct GstRTSPMediaFactoryURI" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI"/>
+ <keyword type="struct" name="struct GstRTSPMediaFactoryURIClass" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURIClass"/>
+ <keyword type="function" name="gst_rtsp_media_new ()" link="GstRTSPMedia.html#gst-rtsp-media-new"/>
+ <keyword type="function" name="gst_rtsp_media_get_element ()" link="GstRTSPMedia.html#gst-rtsp-media-get-element"/>
+ <keyword type="function" name="gst_rtsp_media_take_pipeline ()" link="GstRTSPMedia.html#gst-rtsp-media-take-pipeline"/>
+ <keyword type="function" name="gst_rtsp_media_set_permissions ()" link="GstRTSPMedia.html#gst-rtsp-media-set-permissions"/>
+ <keyword type="function" name="gst_rtsp_media_get_permissions ()" link="GstRTSPMedia.html#gst-rtsp-media-get-permissions"/>
+ <keyword type="function" name="gst_rtsp_media_set_shared ()" link="GstRTSPMedia.html#gst-rtsp-media-set-shared"/>
+ <keyword type="function" name="gst_rtsp_media_is_shared ()" link="GstRTSPMedia.html#gst-rtsp-media-is-shared"/>
+ <keyword type="function" name="gst_rtsp_media_set_reusable ()" link="GstRTSPMedia.html#gst-rtsp-media-set-reusable"/>
+ <keyword type="function" name="gst_rtsp_media_is_reusable ()" link="GstRTSPMedia.html#gst-rtsp-media-is-reusable"/>
+ <keyword type="function" name="gst_rtsp_media_set_profiles ()" link="GstRTSPMedia.html#gst-rtsp-media-set-profiles"/>
+ <keyword type="function" name="gst_rtsp_media_get_profiles ()" link="GstRTSPMedia.html#gst-rtsp-media-get-profiles"/>
+ <keyword type="function" name="gst_rtsp_media_set_protocols ()" link="GstRTSPMedia.html#gst-rtsp-media-set-protocols"/>
+ <keyword type="function" name="gst_rtsp_media_get_protocols ()" link="GstRTSPMedia.html#gst-rtsp-media-get-protocols"/>
+ <keyword type="function" name="gst_rtsp_media_set_eos_shutdown ()" link="GstRTSPMedia.html#gst-rtsp-media-set-eos-shutdown"/>
+ <keyword type="function" name="gst_rtsp_media_is_eos_shutdown ()" link="GstRTSPMedia.html#gst-rtsp-media-is-eos-shutdown"/>
+ <keyword type="function" name="gst_rtsp_media_set_address_pool ()" link="GstRTSPMedia.html#gst-rtsp-media-set-address-pool"/>
+ <keyword type="function" name="gst_rtsp_media_get_address_pool ()" link="GstRTSPMedia.html#gst-rtsp-media-get-address-pool"/>
+ <keyword type="function" name="gst_rtsp_media_set_buffer_size ()" link="GstRTSPMedia.html#gst-rtsp-media-set-buffer-size"/>
+ <keyword type="function" name="gst_rtsp_media_get_buffer_size ()" link="GstRTSPMedia.html#gst-rtsp-media-get-buffer-size"/>
+ <keyword type="function" name="gst_rtsp_media_setup_sdp ()" link="GstRTSPMedia.html#gst-rtsp-media-setup-sdp"/>
+ <keyword type="function" name="gst_rtsp_media_prepare ()" link="GstRTSPMedia.html#gst-rtsp-media-prepare"/>
+ <keyword type="function" name="gst_rtsp_media_unprepare ()" link="GstRTSPMedia.html#gst-rtsp-media-unprepare"/>
+ <keyword type="function" name="gst_rtsp_media_get_status ()" link="GstRTSPMedia.html#gst-rtsp-media-get-status"/>
+ <keyword type="function" name="gst_rtsp_media_set_suspend_mode ()" link="GstRTSPMedia.html#gst-rtsp-media-set-suspend-mode"/>
+ <keyword type="function" name="gst_rtsp_media_get_suspend_mode ()" link="GstRTSPMedia.html#gst-rtsp-media-get-suspend-mode"/>
+ <keyword type="function" name="gst_rtsp_media_suspend ()" link="GstRTSPMedia.html#gst-rtsp-media-suspend"/>
+ <keyword type="function" name="gst_rtsp_media_unsuspend ()" link="GstRTSPMedia.html#gst-rtsp-media-unsuspend"/>
+ <keyword type="function" name="gst_rtsp_media_collect_streams ()" link="GstRTSPMedia.html#gst-rtsp-media-collect-streams"/>
+ <keyword type="function" name="gst_rtsp_media_create_stream ()" link="GstRTSPMedia.html#gst-rtsp-media-create-stream"/>
+ <keyword type="function" name="gst_rtsp_media_n_streams ()" link="GstRTSPMedia.html#gst-rtsp-media-n-streams"/>
+ <keyword type="function" name="gst_rtsp_media_get_stream ()" link="GstRTSPMedia.html#gst-rtsp-media-get-stream"/>
+ <keyword type="function" name="gst_rtsp_media_find_stream ()" link="GstRTSPMedia.html#gst-rtsp-media-find-stream"/>
+ <keyword type="function" name="gst_rtsp_media_seek ()" link="GstRTSPMedia.html#gst-rtsp-media-seek"/>
+ <keyword type="function" name="gst_rtsp_media_get_range_string ()" link="GstRTSPMedia.html#gst-rtsp-media-get-range-string"/>
+ <keyword type="function" name="gst_rtsp_media_set_state ()" link="GstRTSPMedia.html#gst-rtsp-media-set-state"/>
+ <keyword type="function" name="gst_rtsp_media_set_pipeline_state ()" link="GstRTSPMedia.html#gst-rtsp-media-set-pipeline-state"/>
+ <keyword type="function" name="gst_rtsp_media_get_clock ()" link="GstRTSPMedia.html#gst-rtsp-media-get-clock"/>
+ <keyword type="function" name="gst_rtsp_media_get_base_time ()" link="GstRTSPMedia.html#gst-rtsp-media-get-base-time"/>
+ <keyword type="function" name="gst_rtsp_media_use_time_provider ()" link="GstRTSPMedia.html#gst-rtsp-media-use-time-provider"/>
+ <keyword type="function" name="gst_rtsp_media_is_time_provider ()" link="GstRTSPMedia.html#gst-rtsp-media-is-time-provider"/>
+ <keyword type="function" name="gst_rtsp_media_get_time_provider ()" link="GstRTSPMedia.html#gst-rtsp-media-get-time-provider"/>
+ <keyword type="struct" name="struct GstRTSPMedia" link="GstRTSPMedia.html#GstRTSPMedia-struct"/>
+ <keyword type="struct" name="struct GstRTSPMediaClass" link="GstRTSPMedia.html#GstRTSPMediaClass"/>
+ <keyword type="enum" name="enum GstRTSPMediaStatus" link="GstRTSPMedia.html#GstRTSPMediaStatus"/>
+ <keyword type="enum" name="enum GstRTSPSuspendMode" link="GstRTSPMedia.html#GstRTSPSuspendMode"/>
+ <keyword type="property" name="The “buffer-size” property" link="GstRTSPMedia.html#GstRTSPMedia--buffer-size"/>
+ <keyword type="property" name="The “element” property" link="GstRTSPMedia.html#GstRTSPMedia--element"/>
+ <keyword type="property" name="The “eos-shutdown” property" link="GstRTSPMedia.html#GstRTSPMedia--eos-shutdown"/>
+ <keyword type="property" name="The “profiles” property" link="GstRTSPMedia.html#GstRTSPMedia--profiles"/>
+ <keyword type="property" name="The “protocols” property" link="GstRTSPMedia.html#GstRTSPMedia--protocols"/>
+ <keyword type="property" name="The “reusable” property" link="GstRTSPMedia.html#GstRTSPMedia--reusable"/>
+ <keyword type="property" name="The “shared” property" link="GstRTSPMedia.html#GstRTSPMedia--shared"/>
+ <keyword type="property" name="The “suspend-mode” property" link="GstRTSPMedia.html#GstRTSPMedia--suspend-mode"/>
+ <keyword type="property" name="The “time-provider” property" link="GstRTSPMedia.html#GstRTSPMedia--time-provider"/>
+ <keyword type="signal" name="The “new-state” signal" link="GstRTSPMedia.html#GstRTSPMedia-new-state"/>
+ <keyword type="signal" name="The “new-stream” signal" link="GstRTSPMedia.html#GstRTSPMedia-new-stream"/>
+ <keyword type="signal" name="The “prepared” signal" link="GstRTSPMedia.html#GstRTSPMedia-prepared"/>
+ <keyword type="signal" name="The “removed-stream” signal" link="GstRTSPMedia.html#GstRTSPMedia-removed-stream"/>
+ <keyword type="signal" name="The “target-state” signal" link="GstRTSPMedia.html#GstRTSPMedia-target-state"/>
+ <keyword type="signal" name="The “unprepared” signal" link="GstRTSPMedia.html#GstRTSPMedia-unprepared"/>
+ <keyword type="function" name="gst_rtsp_stream_new ()" link="GstRTSPStream.html#gst-rtsp-stream-new"/>
+ <keyword type="function" name="gst_rtsp_stream_get_index ()" link="GstRTSPStream.html#gst-rtsp-stream-get-index"/>
+ <keyword type="function" name="gst_rtsp_stream_get_srcpad ()" link="GstRTSPStream.html#gst-rtsp-stream-get-srcpad"/>
+ <keyword type="function" name="gst_rtsp_stream_get_control ()" link="GstRTSPStream.html#gst-rtsp-stream-get-control"/>
+ <keyword type="function" name="gst_rtsp_stream_set_control ()" link="GstRTSPStream.html#gst-rtsp-stream-set-control"/>
+ <keyword type="function" name="gst_rtsp_stream_has_control ()" link="GstRTSPStream.html#gst-rtsp-stream-has-control"/>
+ <keyword type="function" name="gst_rtsp_stream_get_mtu ()" link="GstRTSPStream.html#gst-rtsp-stream-get-mtu"/>
+ <keyword type="function" name="gst_rtsp_stream_set_mtu ()" link="GstRTSPStream.html#gst-rtsp-stream-set-mtu"/>
+ <keyword type="function" name="gst_rtsp_stream_get_dscp_qos ()" link="GstRTSPStream.html#gst-rtsp-stream-get-dscp-qos"/>
+ <keyword type="function" name="gst_rtsp_stream_set_dscp_qos ()" link="GstRTSPStream.html#gst-rtsp-stream-set-dscp-qos"/>
+ <keyword type="function" name="gst_rtsp_stream_set_profiles ()" link="GstRTSPStream.html#gst-rtsp-stream-set-profiles"/>
+ <keyword type="function" name="gst_rtsp_stream_get_profiles ()" link="GstRTSPStream.html#gst-rtsp-stream-get-profiles"/>
+ <keyword type="function" name="gst_rtsp_stream_get_protocols ()" link="GstRTSPStream.html#gst-rtsp-stream-get-protocols"/>
+ <keyword type="function" name="gst_rtsp_stream_set_protocols ()" link="GstRTSPStream.html#gst-rtsp-stream-set-protocols"/>
+ <keyword type="function" name="gst_rtsp_stream_is_transport_supported ()" link="GstRTSPStream.html#gst-rtsp-stream-is-transport-supported"/>
+ <keyword type="function" name="gst_rtsp_stream_get_address_pool ()" link="GstRTSPStream.html#gst-rtsp-stream-get-address-pool"/>
+ <keyword type="function" name="gst_rtsp_stream_set_address_pool ()" link="GstRTSPStream.html#gst-rtsp-stream-set-address-pool"/>
+ <keyword type="function" name="gst_rtsp_stream_reserve_address ()" link="GstRTSPStream.html#gst-rtsp-stream-reserve-address"/>
+ <keyword type="function" name="gst_rtsp_stream_join_bin ()" link="GstRTSPStream.html#gst-rtsp-stream-join-bin"/>
+ <keyword type="function" name="gst_rtsp_stream_leave_bin ()" link="GstRTSPStream.html#gst-rtsp-stream-leave-bin"/>
+ <keyword type="function" name="gst_rtsp_stream_get_server_port ()" link="GstRTSPStream.html#gst-rtsp-stream-get-server-port"/>
+ <keyword type="function" name="gst_rtsp_stream_get_multicast_address ()" link="GstRTSPStream.html#gst-rtsp-stream-get-multicast-address"/>
+ <keyword type="function" name="gst_rtsp_stream_get_rtpsession ()" link="GstRTSPStream.html#gst-rtsp-stream-get-rtpsession"/>
+ <keyword type="function" name="gst_rtsp_stream_get_ssrc ()" link="GstRTSPStream.html#gst-rtsp-stream-get-ssrc"/>
+ <keyword type="function" name="gst_rtsp_stream_get_rtpinfo ()" link="GstRTSPStream.html#gst-rtsp-stream-get-rtpinfo"/>
+ <keyword type="function" name="gst_rtsp_stream_get_caps ()" link="GstRTSPStream.html#gst-rtsp-stream-get-caps"/>
+ <keyword type="function" name="gst_rtsp_stream_get_pt ()" link="GstRTSPStream.html#gst-rtsp-stream-get-pt"/>
+ <keyword type="function" name="gst_rtsp_stream_recv_rtcp ()" link="GstRTSPStream.html#gst-rtsp-stream-recv-rtcp"/>
+ <keyword type="function" name="gst_rtsp_stream_recv_rtp ()" link="GstRTSPStream.html#gst-rtsp-stream-recv-rtp"/>
+ <keyword type="function" name="gst_rtsp_stream_add_transport ()" link="GstRTSPStream.html#gst-rtsp-stream-add-transport"/>
+ <keyword type="function" name="gst_rtsp_stream_remove_transport ()" link="GstRTSPStream.html#gst-rtsp-stream-remove-transport"/>
+ <keyword type="function" name="gst_rtsp_stream_get_rtp_socket ()" link="GstRTSPStream.html#gst-rtsp-stream-get-rtp-socket"/>
+ <keyword type="function" name="gst_rtsp_stream_get_rtcp_socket ()" link="GstRTSPStream.html#gst-rtsp-stream-get-rtcp-socket"/>
+ <keyword type="function" name="gst_rtsp_stream_set_blocked ()" link="GstRTSPStream.html#gst-rtsp-stream-set-blocked"/>
+ <keyword type="function" name="gst_rtsp_stream_is_blocking ()" link="GstRTSPStream.html#gst-rtsp-stream-is-blocking"/>
+ <keyword type="function" name="gst_rtsp_stream_update_crypto ()" link="GstRTSPStream.html#gst-rtsp-stream-update-crypto"/>
+ <keyword type="function" name="GstRTSPStreamTransportFilterFunc ()" link="GstRTSPStream.html#GstRTSPStreamTransportFilterFunc"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_filter ()" link="GstRTSPStream.html#gst-rtsp-stream-transport-filter"/>
+ <keyword type="struct" name="struct GstRTSPStream" link="GstRTSPStream.html#GstRTSPStream-struct"/>
+ <keyword type="struct" name="GstRTSPStreamClass" link="GstRTSPStream.html#GstRTSPStreamClass"/>
+ <keyword type="function" name="gst_rtsp_session_pool_new ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-new"/>
+ <keyword type="function" name="gst_rtsp_session_pool_get_max_sessions ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-max-sessions"/>
+ <keyword type="function" name="gst_rtsp_session_pool_set_max_sessions ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-set-max-sessions"/>
+ <keyword type="function" name="gst_rtsp_session_pool_get_n_sessions ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-n-sessions"/>
+ <keyword type="function" name="gst_rtsp_session_pool_create ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-create"/>
+ <keyword type="function" name="gst_rtsp_session_pool_find ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-find"/>
+ <keyword type="function" name="gst_rtsp_session_pool_remove ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-remove"/>
+ <keyword type="function" name="gst_rtsp_session_pool_cleanup ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-cleanup"/>
+ <keyword type="function" name="GstRTSPSessionPoolFunc ()" link="GstRTSPSessionPool.html#GstRTSPSessionPoolFunc"/>
+ <keyword type="function" name="gst_rtsp_session_pool_create_watch ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-create-watch"/>
+ <keyword type="function" name="GstRTSPSessionPoolFilterFunc ()" link="GstRTSPSessionPool.html#GstRTSPSessionPoolFilterFunc"/>
+ <keyword type="function" name="gst_rtsp_session_pool_filter ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter"/>
+ <keyword type="struct" name="struct GstRTSPSessionPool" link="GstRTSPSessionPool.html#GstRTSPSessionPool-struct"/>
+ <keyword type="struct" name="struct GstRTSPSessionPoolClass" link="GstRTSPSessionPool.html#GstRTSPSessionPoolClass"/>
+ <keyword type="property" name="The “max-sessions” property" link="GstRTSPSessionPool.html#GstRTSPSessionPool--max-sessions"/>
+ <keyword type="signal" name="The “session-removed” signal" link="GstRTSPSessionPool.html#GstRTSPSessionPool-session-removed"/>
+ <keyword type="function" name="gst_rtsp_session_new ()" link="GstRTSPSession.html#gst-rtsp-session-new"/>
+ <keyword type="function" name="gst_rtsp_session_get_sessionid ()" link="GstRTSPSession.html#gst-rtsp-session-get-sessionid"/>
+ <keyword type="function" name="gst_rtsp_session_get_header ()" link="GstRTSPSession.html#gst-rtsp-session-get-header"/>
+ <keyword type="function" name="gst_rtsp_session_set_timeout ()" link="GstRTSPSession.html#gst-rtsp-session-set-timeout"/>
+ <keyword type="function" name="gst_rtsp_session_get_timeout ()" link="GstRTSPSession.html#gst-rtsp-session-get-timeout"/>
+ <keyword type="function" name="gst_rtsp_session_touch ()" link="GstRTSPSession.html#gst-rtsp-session-touch"/>
+ <keyword type="function" name="gst_rtsp_session_prevent_expire ()" link="GstRTSPSession.html#gst-rtsp-session-prevent-expire"/>
+ <keyword type="function" name="gst_rtsp_session_allow_expire ()" link="GstRTSPSession.html#gst-rtsp-session-allow-expire"/>
+ <keyword type="function" name="gst_rtsp_session_next_timeout ()" link="GstRTSPSession.html#gst-rtsp-session-next-timeout"/>
+ <keyword type="function" name="gst_rtsp_session_is_expired ()" link="GstRTSPSession.html#gst-rtsp-session-is-expired"/>
+ <keyword type="function" name="gst_rtsp_session_manage_media ()" link="GstRTSPSession.html#gst-rtsp-session-manage-media"/>
+ <keyword type="function" name="gst_rtsp_session_release_media ()" link="GstRTSPSession.html#gst-rtsp-session-release-media"/>
+ <keyword type="function" name="gst_rtsp_session_get_media ()" link="GstRTSPSession.html#gst-rtsp-session-get-media"/>
+ <keyword type="function" name="GstRTSPSessionFilterFunc ()" link="GstRTSPSession.html#GstRTSPSessionFilterFunc"/>
+ <keyword type="function" name="gst_rtsp_session_filter ()" link="GstRTSPSession.html#gst-rtsp-session-filter"/>
+ <keyword type="struct" name="struct GstRTSPSession" link="GstRTSPSession.html#GstRTSPSession-struct"/>
+ <keyword type="struct" name="struct GstRTSPSessionClass" link="GstRTSPSession.html#GstRTSPSessionClass"/>
+ <keyword type="enum" name="enum GstRTSPFilterResult" link="GstRTSPSession.html#GstRTSPFilterResult"/>
+ <keyword type="property" name="The “sessionid” property" link="GstRTSPSession.html#GstRTSPSession--sessionid"/>
+ <keyword type="property" name="The “timeout” property" link="GstRTSPSession.html#GstRTSPSession--timeout"/>
+ <keyword type="property" name="The “timeout-always-visible” property" link="GstRTSPSession.html#GstRTSPSession--timeout-always-visible"/>
+ <keyword type="function" name="gst_rtsp_session_media_new ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-new"/>
+ <keyword type="function" name="gst_rtsp_session_media_matches ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-matches"/>
+ <keyword type="function" name="gst_rtsp_session_media_get_media ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-media"/>
+ <keyword type="function" name="gst_rtsp_session_media_get_base_time ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-base-time"/>
+ <keyword type="function" name="gst_rtsp_session_media_get_rtpinfo ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtpinfo"/>
+ <keyword type="function" name="gst_rtsp_session_media_set_state ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-state"/>
+ <keyword type="function" name="gst_rtsp_session_media_get_rtsp_state ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtsp-state"/>
+ <keyword type="function" name="gst_rtsp_session_media_set_rtsp_state ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-rtsp-state"/>
+ <keyword type="function" name="gst_rtsp_session_media_get_transport ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-transport"/>
+ <keyword type="function" name="gst_rtsp_session_media_set_transport ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-transport"/>
+ <keyword type="function" name="gst_rtsp_session_media_alloc_channels ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-alloc-channels"/>
+ <keyword type="struct" name="struct GstRTSPSessionMedia" link="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia"/>
+ <keyword type="struct" name="struct GstRTSPSessionMediaClass" link="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMediaClass"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_new ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-new"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_get_stream ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-stream"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_get_transport ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-transport"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_set_transport ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-transport"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_get_url ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-url"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_set_url ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-url"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_get_rtpinfo ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-rtpinfo"/>
+ <keyword type="function" name="GstRTSPSendFunc ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPSendFunc"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_set_callbacks ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-callbacks"/>
+ <keyword type="function" name="GstRTSPKeepAliveFunc ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPKeepAliveFunc"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_set_keepalive ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-keepalive"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_keep_alive ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-keep-alive"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_set_active ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-active"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_set_timed_out ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-timed-out"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_is_timed_out ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-is-timed-out"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_send_rtcp ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtcp"/>
+ <keyword type="function" name="gst_rtsp_stream_transport_send_rtp ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtp"/>
+ <keyword type="struct" name="struct GstRTSPStreamTransport" link="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport"/>
+ <keyword type="struct" name="struct GstRTSPStreamTransportClass" link="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransportClass"/>
+ <keyword type="function" name="gst_rtsp_sdp_from_media ()" link="gst-rtsp-server-GstRTSPSdp.html#gst-rtsp-sdp-from-media"/>
+ <keyword type="struct" name="GstSDPInfo" link="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo"/>
+ <keyword type="function" name="gst_rtsp_address_copy ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-copy"/>
+ <keyword type="function" name="gst_rtsp_address_free ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free"/>
+ <keyword type="function" name="gst_rtsp_address_pool_new ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-new"/>
+ <keyword type="function" name="gst_rtsp_address_pool_clear ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-clear"/>
+ <keyword type="function" name="gst_rtsp_address_pool_dump ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-dump"/>
+ <keyword type="function" name="gst_rtsp_address_pool_add_range ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range"/>
+ <keyword type="function" name="gst_rtsp_address_pool_has_unicast_addresses ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-has-unicast-addresses"/>
+ <keyword type="function" name="gst_rtsp_address_pool_acquire_address ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-acquire-address"/>
+ <keyword type="function" name="gst_rtsp_address_pool_reserve_address ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-reserve-address"/>
+ <keyword type="macro" name="GST_RTSP_ADDRESS_POOL_ANY_IPV4" link="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV4:CAPS"/>
+ <keyword type="macro" name="GST_RTSP_ADDRESS_POOL_ANY_IPV6" link="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV6:CAPS"/>
+ <keyword type="struct" name="struct GstRTSPAddress" link="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress"/>
+ <keyword type="enum" name="enum GstRTSPAddressFlags" link="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressFlags"/>
+ <keyword type="struct" name="struct GstRTSPAddressPool" link="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool"/>
+ <keyword type="struct" name="struct GstRTSPAddressPoolClass" link="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolClass"/>
+ <keyword type="enum" name="enum GstRTSPAddressPoolResult" link="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolResult"/>
+ <keyword type="function" name="gst_rtsp_thread_new ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-new"/>
+ <keyword type="function" name="gst_rtsp_thread_ref ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-ref"/>
+ <keyword type="function" name="gst_rtsp_thread_unref ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-unref"/>
+ <keyword type="function" name="gst_rtsp_thread_reuse ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-reuse"/>
+ <keyword type="function" name="gst_rtsp_thread_stop ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-stop"/>
+ <keyword type="function" name="gst_rtsp_thread_pool_new ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-new"/>
+ <keyword type="function" name="gst_rtsp_thread_pool_get_max_threads ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-max-threads"/>
+ <keyword type="function" name="gst_rtsp_thread_pool_set_max_threads ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-set-max-threads"/>
+ <keyword type="function" name="gst_rtsp_thread_pool_get_thread ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-thread"/>
+ <keyword type="function" name="gst_rtsp_thread_pool_cleanup ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-cleanup"/>
+ <keyword type="enum" name="enum GstRTSPThreadType" link="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType"/>
+ <keyword type="struct" name="struct GstRTSPThread" link="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread"/>
+ <keyword type="struct" name="struct GstRTSPThreadPool" link="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool"/>
+ <keyword type="struct" name="struct GstRTSPThreadPoolClass" link="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPoolClass"/>
+ <keyword type="function" name="gst_rtsp_auth_new ()" link="GstRTSPAuth.html#gst-rtsp-auth-new"/>
+ <keyword type="function" name="gst_rtsp_auth_get_tls_certificate ()" link="GstRTSPAuth.html#gst-rtsp-auth-get-tls-certificate"/>
+ <keyword type="function" name="gst_rtsp_auth_set_tls_certificate ()" link="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate"/>
+ <keyword type="function" name="gst_rtsp_auth_make_basic ()" link="GstRTSPAuth.html#gst-rtsp-auth-make-basic"/>
+ <keyword type="function" name="gst_rtsp_auth_add_basic ()" link="GstRTSPAuth.html#gst-rtsp-auth-add-basic"/>
+ <keyword type="function" name="gst_rtsp_auth_remove_basic ()" link="GstRTSPAuth.html#gst-rtsp-auth-remove-basic"/>
+ <keyword type="function" name="gst_rtsp_auth_check ()" link="GstRTSPAuth.html#gst-rtsp-auth-check"/>
+ <keyword type="function" name="gst_rtsp_auth_get_default_token ()" link="GstRTSPAuth.html#gst-rtsp-auth-get-default-token"/>
+ <keyword type="function" name="gst_rtsp_auth_set_default_token ()" link="GstRTSPAuth.html#gst-rtsp-auth-set-default-token"/>
+ <keyword type="struct" name="struct GstRTSPAuth" link="GstRTSPAuth.html#GstRTSPAuth-struct"/>
+ <keyword type="struct" name="struct GstRTSPAuthClass" link="GstRTSPAuth.html#GstRTSPAuthClass"/>
+ <keyword type="macro" name="GST_RTSP_AUTH_CHECK_CONNECT" link="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-CONNECT:CAPS"/>
+ <keyword type="macro" name="GST_RTSP_AUTH_CHECK_URL" link="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-URL:CAPS"/>
+ <keyword type="macro" name="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS" link="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-ACCESS:CAPS"/>
+ <keyword type="macro" name="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT" link="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-CONSTRUCT:CAPS"/>
+ <keyword type="macro" name="GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS" link="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-TRANSPORT-CLIENT-SETTINGS:CAPS"/>
+ <keyword type="macro" name="GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE" link="GstRTSPAuth.html#GST-RTSP-TOKEN-MEDIA-FACTORY-ROLE:CAPS"/>
+ <keyword type="macro" name="GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS" link="GstRTSPAuth.html#GST-RTSP-TOKEN-TRANSPORT-CLIENT-SETTINGS:CAPS"/>
+ <keyword type="macro" name="GST_RTSP_PERM_MEDIA_FACTORY_ACCESS" link="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-ACCESS:CAPS"/>
+ <keyword type="macro" name="GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT" link="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-CONSTRUCT:CAPS"/>
+ <keyword type="function" name="gst_rtsp_token_new_empty ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-empty"/>
+ <keyword type="function" name="gst_rtsp_token_new ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new"/>
+ <keyword type="function" name="gst_rtsp_token_new_valist ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-valist"/>
+ <keyword type="function" name="gst_rtsp_token_ref ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-ref"/>
+ <keyword type="function" name="gst_rtsp_token_unref ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-unref"/>
+ <keyword type="function" name="gst_rtsp_token_get_structure ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-structure"/>
+ <keyword type="function" name="gst_rtsp_token_writable_structure ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-writable-structure"/>
+ <keyword type="function" name="gst_rtsp_token_get_string ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-string"/>
+ <keyword type="function" name="gst_rtsp_token_is_allowed ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-is-allowed"/>
+ <keyword type="struct" name="struct GstRTSPToken" link="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken"/>
+ <keyword type="function" name="gst_rtsp_permissions_new ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-new"/>
+ <keyword type="function" name="gst_rtsp_permissions_ref ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-ref"/>
+ <keyword type="function" name="gst_rtsp_permissions_unref ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-unref"/>
+ <keyword type="function" name="gst_rtsp_permissions_add_role ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role"/>
+ <keyword type="function" name="gst_rtsp_permissions_add_role_valist ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role-valist"/>
+ <keyword type="function" name="gst_rtsp_permissions_remove_role ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-remove-role"/>
+ <keyword type="function" name="gst_rtsp_permissions_get_role ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-get-role"/>
+ <keyword type="function" name="gst_rtsp_permissions_is_allowed ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-is-allowed"/>
+ <keyword type="struct" name="struct GstRTSPPermissions" link="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions"/>
+ <keyword type="function" name="gst_rtsp_params_get ()" link="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-get"/>
+ <keyword type="function" name="gst_rtsp_params_set ()" link="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-set"/>
+ </functions>
+</book>
--- /dev/null
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+<div class="refentry">
+<a name="gst-rtsp-server-GstRTSPAddressPool"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPAddressPool.top_of_page"></a>GstRTSPAddressPool</span></h2>
+<p>GstRTSPAddressPool — A pool of network addresses</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPAddressPool.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-copy" title="gst_rtsp_address_copy ()">gst_rtsp_address_copy</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free" title="gst_rtsp_address_free ()">gst_rtsp_address_free</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-new" title="gst_rtsp_address_pool_new ()">gst_rtsp_address_pool_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-clear" title="gst_rtsp_address_pool_clear ()">gst_rtsp_address_pool_clear</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-dump" title="gst_rtsp_address_pool_dump ()">gst_rtsp_address_pool_dump</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range" title="gst_rtsp_address_pool_add_range ()">gst_rtsp_address_pool_add_range</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-has-unicast-addresses" title="gst_rtsp_address_pool_has_unicast_addresses ()">gst_rtsp_address_pool_has_unicast_addresses</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-acquire-address" title="gst_rtsp_address_pool_acquire_address ()">gst_rtsp_address_pool_acquire_address</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolResult" title="enum GstRTSPAddressPoolResult"><span class="returnvalue">GstRTSPAddressPoolResult</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-reserve-address" title="gst_rtsp_address_pool_reserve_address ()">gst_rtsp_address_pool_reserve_address</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPAddressPool.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV4:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV4">GST_RTSP_ADDRESS_POOL_ANY_IPV4</a></td>
+</tr>
+<tr>
+<td class="define_keyword">#define</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV6:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV6">GST_RTSP_ADDRESS_POOL_ANY_IPV6</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress">GstRTSPAddress</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">enum</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressFlags" title="enum GstRTSPAddressFlags">GstRTSPAddressFlags</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool">GstRTSPAddressPool</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolClass" title="struct GstRTSPAddressPoolClass">GstRTSPAddressPoolClass</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">enum</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolResult" title="enum GstRTSPAddressPoolResult">GstRTSPAddressPoolResult</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPAddressPool.description"></a><h2>Description</h2>
+<p>The <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> is an object that maintains a collection of network
+addresses. It is used to allocate server ports and server multicast addresses
+but also to reserve client provided destination addresses.</p>
+<p>A range of addresses can be added with <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range" title="gst_rtsp_address_pool_add_range ()"><code class="function">gst_rtsp_address_pool_add_range()</code></a>.
+Both multicast and unicast addresses can be added.</p>
+<p>With <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-acquire-address" title="gst_rtsp_address_pool_acquire_address ()"><code class="function">gst_rtsp_address_pool_acquire_address()</code></a> an unused address and port range
+can be acquired from the pool. With <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-reserve-address" title="gst_rtsp_address_pool_reserve_address ()"><code class="function">gst_rtsp_address_pool_reserve_address()</code></a> a
+specific address can be retrieved. Both methods return a boxed
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> that should be freed with <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free" title="gst_rtsp_address_free ()"><code class="function">gst_rtsp_address_free()</code></a> after
+usage, which brings the address back into the pool.</p>
+<p>Last reviewed on 2013-07-16 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPAddressPool.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-address-copy"></a><h3>gst_rtsp_address_copy ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
+gst_rtsp_address_copy (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> *addr</code></em>);</pre>
+<p>Make a copy of <em class="parameter"><code>addr</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.14.6.2.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>addr</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.14.6.2.6"></a><h4>Returns</h4>
+<p> a copy of <em class="parameter"><code>addr</code></em>
+.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-address-free"></a><h3>gst_rtsp_address_free ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_address_free (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> *addr</code></em>);</pre>
+<p>Free <em class="parameter"><code>addr</code></em>
+ and releasing it back into the pool when owned by a
+pool.</p>
+<div class="refsect3">
+<a name="id-1.2.14.6.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>addr</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-address-pool-new"></a><h3>gst_rtsp_address_pool_new ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
+gst_rtsp_address_pool_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Make a new <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.14.6.4.5"></a><h4>Returns</h4>
+<p> a new <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-address-pool-clear"></a><h3>gst_rtsp_address_pool_clear ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_address_pool_clear (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
+<p>Clear all addresses in <em class="parameter"><code>pool</code></em>
+. There should be no outstanding
+allocations.</p>
+<div class="refsect3">
+<a name="id-1.2.14.6.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-address-pool-dump"></a><h3>gst_rtsp_address_pool_dump ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_address_pool_dump (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
+<p>Dump the free and allocated addresses to stdout.</p>
+<div class="refsect3">
+<a name="id-1.2.14.6.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-address-pool-add-range"></a><h3>gst_rtsp_address_pool_add_range ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_address_pool_add_range (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *min_address</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *max_address</code></em>,
+ <em class="parameter"><code><span class="type">guint16</span> min_port</code></em>,
+ <em class="parameter"><code><span class="type">guint16</span> max_port</code></em>,
+ <em class="parameter"><code><span class="type">guint8</span> ttl</code></em>);</pre>
+<p>Adds the addresses from <em class="parameter"><code>min_addess</code></em>
+ to <em class="parameter"><code>max_address</code></em>
+ (inclusive)
+to <em class="parameter"><code>pool</code></em>
+. The valid port range for the addresses will be from <em class="parameter"><code>min_port</code></em>
+ to
+<em class="parameter"><code>max_port</code></em>
+ inclusive.</p>
+<p>When <em class="parameter"><code>ttl</code></em>
+ is 0, <em class="parameter"><code>min_address</code></em>
+ and <em class="parameter"><code>max_address</code></em>
+ should be unicast addresses.
+<em class="parameter"><code>min_address</code></em>
+ and <em class="parameter"><code>max_address</code></em>
+ can be set to
+<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV4:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV4"><span class="type">GST_RTSP_ADDRESS_POOL_ANY_IPV4</span></a> or <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV6:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV6"><span class="type">GST_RTSP_ADDRESS_POOL_ANY_IPV6</span></a> to bind
+to all available IPv4 or IPv6 addresses.</p>
+<p>When <em class="parameter"><code>ttl</code></em>
+ > 0, <em class="parameter"><code>min_address</code></em>
+ and <em class="parameter"><code>max_address</code></em>
+ should be multicast addresses.</p>
+<div class="refsect3">
+<a name="id-1.2.14.6.7.7"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>min_address</p></td>
+<td class="parameter_description"><p>a minimum address to add</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>max_address</p></td>
+<td class="parameter_description"><p>a maximum address to add</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>min_port</p></td>
+<td class="parameter_description"><p>the minimum port</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>max_port</p></td>
+<td class="parameter_description"><p>the maximum port</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>ttl</p></td>
+<td class="parameter_description"><p>a TTL or 0 for unicast addresses</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.14.6.7.8"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the addresses could be added.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-address-pool-has-unicast-addresses"></a><h3>gst_rtsp_address_pool_has_unicast_addresses ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_address_pool_has_unicast_addresses
+ (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
+<p>Used to know if the pool includes any unicast addresses.</p>
+<div class="refsect3">
+<a name="id-1.2.14.6.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.14.6.8.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the pool includes any unicast addresses, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#FALSE:CAPS"><code class="literal">FALSE</code></a> otherwise</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-address-pool-acquire-address"></a><h3>gst_rtsp_address_pool_acquire_address ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
+gst_rtsp_address_pool_acquire_address (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressFlags" title="enum GstRTSPAddressFlags"><span class="type">GstRTSPAddressFlags</span></a> flags</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> n_ports</code></em>);</pre>
+<p>Take an address and ports from <em class="parameter"><code>pool</code></em>
+. <em class="parameter"><code>flags</code></em>
+ can be used to control the
+allocation. <em class="parameter"><code>n_ports</code></em>
+ consecutive ports will be allocated of which the first
+one can be found in <em class="parameter"><code>port</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.14.6.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>flags</p></td>
+<td class="parameter_description"><p>flags</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>n_ports</p></td>
+<td class="parameter_description"><p>the amount of ports</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.14.6.9.6"></a><h4>Returns</h4>
+<p> a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> that should be freed with
+gst_rtsp_address_free after use or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when no address could be
+acquired. </p>
+<p><span class="annotation">[<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-address-pool-reserve-address"></a><h3>gst_rtsp_address_pool_reserve_address ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolResult" title="enum GstRTSPAddressPoolResult"><span class="returnvalue">GstRTSPAddressPoolResult</span></a>
+gst_rtsp_address_pool_reserve_address (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *ip_address</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> port</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> n_ports</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> ttl</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> **address</code></em>);</pre>
+<p>Take a specific address and ports from <em class="parameter"><code>pool</code></em>
+. <em class="parameter"><code>n_ports</code></em>
+ consecutive
+ports will be allocated of which the first one can be found in
+<em class="parameter"><code>port</code></em>
+.</p>
+<p>If <em class="parameter"><code>ttl</code></em>
+ is 0, <em class="parameter"><code>address</code></em>
+ should be a unicast address. If <em class="parameter"><code>ttl</code></em>
+ > 0, <em class="parameter"><code>address</code></em>
+
+should be a valid multicast address.</p>
+<div class="refsect3">
+<a name="id-1.2.14.6.10.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>ip_address</p></td>
+<td class="parameter_description"><p>The IP address to reserve</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>port</p></td>
+<td class="parameter_description"><p>The first port to reserve</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>n_ports</p></td>
+<td class="parameter_description"><p>The number of ports</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>ttl</p></td>
+<td class="parameter_description"><p>The requested ttl</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>address</p></td>
+<td class="parameter_description"><p> storage for a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.14.6.10.7"></a><h4>Returns</h4>
+<p> <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-OK:CAPS"><span class="type">GST_RTSP_ADDRESS_POOL_OK</span></a> if an address was reserved. The address
+is returned in <em class="parameter"><code>address</code></em>
+and should be freed with gst_rtsp_address_free
+after use.</p>
+<p></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPAddressPool.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GST-RTSP-ADDRESS-POOL-ANY-IPV4:CAPS"></a><h3>GST_RTSP_ADDRESS_POOL_ANY_IPV4</h3>
+<pre class="programlisting">#define GST_RTSP_ADDRESS_POOL_ANY_IPV4 "0.0.0.0"
+</pre>
+<p>Used with <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range" title="gst_rtsp_address_pool_add_range ()"><code class="function">gst_rtsp_address_pool_add_range()</code></a> to bind to all
+IPv4 addresses</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GST-RTSP-ADDRESS-POOL-ANY-IPV6:CAPS"></a><h3>GST_RTSP_ADDRESS_POOL_ANY_IPV6</h3>
+<pre class="programlisting">#define GST_RTSP_ADDRESS_POOL_ANY_IPV6 "::"
+</pre>
+<p>Used with <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range" title="gst_rtsp_address_pool_add_range ()"><code class="function">gst_rtsp_address_pool_add_range()</code></a> to bind to all
+IPv6 addresses</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPAddress"></a><h3>struct GstRTSPAddress</h3>
+<pre class="programlisting">struct GstRTSPAddress {
+ GstRTSPAddressPool *pool;
+
+ gchar *address;
+ guint16 port;
+ gint n_ports;
+ guint8 ttl;
+};
+</pre>
+<p>An address</p>
+<div class="refsect3">
+<a name="id-1.2.14.7.4.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *<em class="structfield"><code><a name="GstRTSPAddress.pool"></a>pool</code></em>;</p></td>
+<td class="struct_member_description"><p>the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> owner of this address</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *<em class="structfield"><code><a name="GstRTSPAddress.address"></a>address</code></em>;</p></td>
+<td class="struct_member_description"><p>the address</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><span class="type">guint16</span> <em class="structfield"><code><a name="GstRTSPAddress.port"></a>port</code></em>;</p></td>
+<td class="struct_member_description"><p>the port number</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> <em class="structfield"><code><a name="GstRTSPAddress.n-ports"></a>n_ports</code></em>;</p></td>
+<td class="struct_member_description"><p>number of ports</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><span class="type">guint8</span> <em class="structfield"><code><a name="GstRTSPAddress.ttl"></a>ttl</code></em>;</p></td>
+<td class="struct_member_description"><p>TTL or 0 for unicast addresses</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPAddressFlags"></a><h3>enum GstRTSPAddressFlags</h3>
+<p>Flags used to control allocation of addresses</p>
+<div class="refsect3">
+<a name="id-1.2.14.7.5.4"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="enum_members_name">
+<col class="enum_members_description">
+<col width="200px" class="enum_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-NONE:CAPS"></a>GST_RTSP_ADDRESS_FLAG_NONE</p></td>
+<td class="enum_member_description">
+<p>no flags</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-IPV4:CAPS"></a>GST_RTSP_ADDRESS_FLAG_IPV4</p></td>
+<td class="enum_member_description">
+<p>an IPv4 address</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-IPV6:CAPS"></a>GST_RTSP_ADDRESS_FLAG_IPV6</p></td>
+<td class="enum_member_description">
+<p>and IPv6 address</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-EVEN-PORT:CAPS"></a>GST_RTSP_ADDRESS_FLAG_EVEN_PORT</p></td>
+<td class="enum_member_description">
+<p>address with an even port</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-MULTICAST:CAPS"></a>GST_RTSP_ADDRESS_FLAG_MULTICAST</p></td>
+<td class="enum_member_description">
+<p>a multicast address</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-UNICAST:CAPS"></a>GST_RTSP_ADDRESS_FLAG_UNICAST</p></td>
+<td class="enum_member_description">
+<p>a unicast address</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPAddressPool"></a><h3>struct GstRTSPAddressPool</h3>
+<pre class="programlisting">struct GstRTSPAddressPool {
+ GObject parent;
+};
+</pre>
+<p>An address pool, all member are private</p>
+<div class="refsect3">
+<a name="id-1.2.14.7.6.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody><tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject"><span class="type">GObject</span></a> <em class="structfield"><code><a name="GstRTSPAddressPool.parent"></a>parent</code></em>;</p></td>
+<td class="struct_member_description"><p>the parent GObject</p></td>
+<td class="struct_member_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPAddressPoolClass"></a><h3>struct GstRTSPAddressPoolClass</h3>
+<pre class="programlisting">struct GstRTSPAddressPoolClass {
+ GObjectClass parent_class;
+};
+</pre>
+<p>Opaque Address pool class.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPAddressPoolResult"></a><h3>enum GstRTSPAddressPoolResult</h3>
+<p>Result codes from RTSP address pool functions.</p>
+<div class="refsect3">
+<a name="id-1.2.14.7.8.4"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="enum_members_name">
+<col class="enum_members_description">
+<col width="200px" class="enum_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-POOL-OK:CAPS"></a>GST_RTSP_ADDRESS_POOL_OK</p></td>
+<td class="enum_member_description">
+<p>no error</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-POOL-EINVAL:CAPS"></a>GST_RTSP_ADDRESS_POOL_EINVAL</p></td>
+<td class="enum_member_description">
+<p>invalid arguments were provided to a function</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-POOL-ERESERVED:CAPS"></a>GST_RTSP_ADDRESS_POOL_ERESERVED</p></td>
+<td class="enum_member_description">
+<p>the addres has already been reserved</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-POOL-ERANGE:CAPS"></a>GST_RTSP_ADDRESS_POOL_ERANGE</p></td>
+<td class="enum_member_description">
+<p>the address is not in the pool</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-POOL-ELAST:CAPS"></a>GST_RTSP_ADDRESS_POOL_ELAST</p></td>
+<td class="enum_member_description">
+<p>last error</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPAddressPool.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
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+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
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+<a name="gst-rtsp-server-GstRTSPContext"></a><div class="titlepage"></div>
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+<td valign="top">
+<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPContext.top_of_page"></a>GstRTSPContext</span></h2>
+<p>GstRTSPContext — A client request context</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPContext.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="returnvalue">GstRTSPContext</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-get-current" title="gst_rtsp_context_get_current ()">gst_rtsp_context_get_current</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-push-current" title="gst_rtsp_context_push_current ()">gst_rtsp_context_push_current</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-pop-current" title="gst_rtsp_context_pop_current ()">gst_rtsp_context_pop_current</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPContext.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody><tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext">GstRTSPContext</a></td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPContext.description"></a><h2>Description</h2>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPContext.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-context-get-current"></a><h3>gst_rtsp_context_get_current ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="returnvalue">GstRTSPContext</span></a> *
+gst_rtsp_context_get_current (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Get the current <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a>. This object is retrieved from the
+current thread that is handling the request for a client.</p>
+<div class="refsect3">
+<a name="id-1.2.3.6.2.5"></a><h4>Returns</h4>
+<p> a <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a></p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-context-push-current"></a><h3>gst_rtsp_context_push_current ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_context_push_current (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *ctx</code></em>);</pre>
+<p>Pushes <em class="parameter"><code>ctx</code></em>
+ onto the context stack. The current
+context can then be received using <a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-get-current" title="gst_rtsp_context_get_current ()"><code class="function">gst_rtsp_context_get_current()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.3.6.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>ctx</p></td>
+<td class="parameter_description"><p>a #<a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-context-pop-current"></a><h3>gst_rtsp_context_pop_current ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_context_pop_current (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *ctx</code></em>);</pre>
+<p>Pops <em class="parameter"><code>ctx</code></em>
+ off the context stack (verifying that <em class="parameter"><code>ctx</code></em>
+
+is on the top of the stack).</p>
+<div class="refsect3">
+<a name="id-1.2.3.6.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>ctx</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPContext.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPContext"></a><h3>struct GstRTSPContext</h3>
+<pre class="programlisting">struct GstRTSPContext {
+ GstRTSPServer *server;
+ GstRTSPConnection *conn;
+ GstRTSPClient *client;
+ GstRTSPMessage *request;
+ GstRTSPUrl *uri;
+ GstRTSPMethod method;
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ GstRTSPSession *session;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPStream *stream;
+ GstRTSPMessage *response;
+};
+</pre>
+<p>Information passed around containing the context of a request.</p>
+<div class="refsect3">
+<a name="id-1.2.3.7.2.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *<em class="structfield"><code><a name="GstRTSPContext.server"></a>server</code></em>;</p></td>
+<td class="struct_member_description"><p>the server</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> *<em class="structfield"><code><a name="GstRTSPContext.conn"></a>conn</code></em>;</p></td>
+<td class="struct_member_description"><p>the connection</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *<em class="structfield"><code><a name="GstRTSPContext.client"></a>client</code></em>;</p></td>
+<td class="struct_member_description"><p>the client</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> *<em class="structfield"><code><a name="GstRTSPContext.request"></a>request</code></em>;</p></td>
+<td class="struct_member_description"><p>the complete request</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a> *<em class="structfield"><code><a name="GstRTSPContext.uri"></a>uri</code></em>;</p></td>
+<td class="struct_member_description"><p>the complete url parsed from <em class="parameter"><code>request</code></em>
+</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPMethod"><span class="type">GstRTSPMethod</span></a> <em class="structfield"><code><a name="GstRTSPContext.method"></a>method</code></em>;</p></td>
+<td class="struct_member_description"><p>the parsed method of <em class="parameter"><code>uri</code></em>
+</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *<em class="structfield"><code><a name="GstRTSPContext.auth"></a>auth</code></em>;</p></td>
+<td class="struct_member_description"><p>the current auth object or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *<em class="structfield"><code><a name="GstRTSPContext.token"></a>token</code></em>;</p></td>
+<td class="struct_member_description"><p>authorisation token</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *<em class="structfield"><code><a name="GstRTSPContext.session"></a>session</code></em>;</p></td>
+<td class="struct_member_description"><p>the session, can be <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *<em class="structfield"><code><a name="GstRTSPContext.sessmedia"></a>sessmedia</code></em>;</p></td>
+<td class="struct_member_description"><p>the session media for the url can be <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *<em class="structfield"><code><a name="GstRTSPContext.factory"></a>factory</code></em>;</p></td>
+<td class="struct_member_description"><p>the media factory for the url, can be <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *<em class="structfield"><code><a name="GstRTSPContext.media"></a>media</code></em>;</p></td>
+<td class="struct_member_description"><p>the media for the url can be <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *<em class="structfield"><code><a name="GstRTSPContext.stream"></a>stream</code></em>;</p></td>
+<td class="struct_member_description"><p>the stream for the url can be <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> *<em class="structfield"><code><a name="GstRTSPContext.response"></a>response</code></em>;</p></td>
+<td class="struct_member_description"><p>the response</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPContext.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a>, <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
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+<title>GStreamer RTSP Server Reference Manual: GstRTSPMediaFactoryURI</title>
+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
+<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
+<link rel="up" href="ch01.html" title="">
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+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#gst-rtsp-server-GstRTSPMediaFactoryURI.description" class="shortcut">Description</a></span>
+</td>
+<td><a accesskey="h" href="index.html"><img src="home.png" width="16" height="16" border="0" alt="Home"></a></td>
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+<td><a accesskey="n" href="GstRTSPMedia.html"><img src="right.png" width="16" height="16" border="0" alt="Next"></a></td>
+</tr></table>
+<div class="refentry">
+<a name="gst-rtsp-server-GstRTSPMediaFactoryURI"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPMediaFactoryURI.top_of_page"></a>GstRTSPMediaFactoryURI</span></h2>
+<p>GstRTSPMediaFactoryURI — A factory for URI sources</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="returnvalue">GstRTSPMediaFactoryURI</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-new" title="gst_rtsp_media_factory_uri_new ()">gst_rtsp_media_factory_uri_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-set-uri" title="gst_rtsp_media_factory_uri_set_uri ()">gst_rtsp_media_factory_uri_set_uri</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-get-uri" title="gst_rtsp_media_factory_uri_get_uri ()">gst_rtsp_media_factory_uri_get_uri</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURIClass" title="struct GstRTSPMediaFactoryURIClass">GstRTSPMediaFactoryURIClass</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.description"></a><h2>Description</h2>
+<p>This specialized <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> constructs media pipelines from a URI,
+given with <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-set-uri" title="gst_rtsp_media_factory_uri_set_uri ()"><code class="function">gst_rtsp_media_factory_uri_set_uri()</code></a>.</p>
+<p>It will automatically demux and payload the different streams found in the
+media at URL.</p>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-uri-new"></a><h3>gst_rtsp_media_factory_uri_new ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="returnvalue">GstRTSPMediaFactoryURI</span></a> *
+gst_rtsp_media_factory_uri_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Create a new <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="type">GstRTSPMediaFactoryURI</span></a> instance.</p>
+<div class="refsect3">
+<a name="id-1.2.6.6.2.5"></a><h4>Returns</h4>
+<p> a new <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="type">GstRTSPMediaFactoryURI</span></a> object. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-uri-set-uri"></a><h3>gst_rtsp_media_factory_uri_set_uri ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_media_factory_uri_set_uri (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="type">GstRTSPMediaFactoryURI</span></a> *factory</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *uri</code></em>);</pre>
+<p>Set the URI of the resource that will be streamed by this factory.</p>
+<div class="refsect3">
+<a name="id-1.2.6.6.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>uri</p></td>
+<td class="parameter_description"><p>the uri the stream</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-media-factory-uri-get-uri"></a><h3>gst_rtsp_media_factory_uri_get_uri ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_media_factory_uri_get_uri (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="type">GstRTSPMediaFactoryURI</span></a> *factory</code></em>);</pre>
+<p>Get the URI that will provide media for this factory.</p>
+<div class="refsect3">
+<a name="id-1.2.6.6.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>factory</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.6.6.4.6"></a><h4>Returns</h4>
+<p> the configured URI. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPMediaFactoryURI"></a><h3>struct GstRTSPMediaFactoryURI</h3>
+<pre class="programlisting">struct GstRTSPMediaFactoryURI {
+ GstRTSPMediaFactory parent;
+};
+</pre>
+<p>A media factory that creates a pipeline to play and uri.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPMediaFactoryURIClass"></a><h3>struct GstRTSPMediaFactoryURIClass</h3>
+<pre class="programlisting">struct GstRTSPMediaFactoryURIClass {
+ GstRTSPMediaFactoryClass parent_class;
+};
+</pre>
+<p>The <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="type">GstRTSPMediaFactoryURI</span></a> class structure.</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>, <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
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+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
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+<link rel="up" href="ch01.html" title="">
+<link rel="prev" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">
+<link rel="next" href="rtsp-server-hierarchy.html" title="Object Hierarchy">
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+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#gst-rtsp-server-GstRTSPParams.description" class="shortcut">Description</a></span>
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+<td><a accesskey="n" href="rtsp-server-hierarchy.html"><img src="right.png" width="16" height="16" border="0" alt="Next"></a></td>
+</tr></table>
+<div class="refentry">
+<a name="gst-rtsp-server-GstRTSPParams"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPParams.top_of_page"></a>GstRTSPParams</span></h2>
+<p>GstRTSPParams — Param get and set implementation</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPParams.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-get" title="gst_rtsp_params_get ()">gst_rtsp_params_get</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-set" title="gst_rtsp_params_set ()">gst_rtsp_params_set</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPParams.description"></a><h2>Description</h2>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPParams.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-params-get"></a><h3>gst_rtsp_params_get ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
+gst_rtsp_params_get (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *ctx</code></em>);</pre>
+<p>Get parameters (not implemented yet)</p>
+<div class="refsect3">
+<a name="id-1.2.19.5.2.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>ctx</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.19.5.2.6"></a><h4>Returns</h4>
+<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="type">GstRTSPResult</span></a></p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-params-set"></a><h3>gst_rtsp_params_set ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
+gst_rtsp_params_set (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *ctx</code></em>);</pre>
+<p>Set parameters (not implemented yet)</p>
+<div class="refsect3">
+<a name="id-1.2.19.5.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>client</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>ctx</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.19.5.3.6"></a><h4>Returns</h4>
+<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="type">GstRTSPResult</span></a></p>
+<p></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPParams.other_details"></a><h2>Types and Values</h2>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPParams.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
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+<title>GStreamer RTSP Server Reference Manual: GstRTSPPermissions</title>
+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
+<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
+<link rel="up" href="ch01.html" title="">
+<link rel="prev" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">
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+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#gst-rtsp-server-GstRTSPPermissions.description" class="shortcut">Description</a></span>
+</td>
+<td><a accesskey="h" href="index.html"><img src="home.png" width="16" height="16" border="0" alt="Home"></a></td>
+<td><a accesskey="u" href="ch01.html"><img src="up.png" width="16" height="16" border="0" alt="Up"></a></td>
+<td><a accesskey="p" href="gst-rtsp-server-GstRTSPToken.html"><img src="left.png" width="16" height="16" border="0" alt="Prev"></a></td>
+<td><a accesskey="n" href="gst-rtsp-server-GstRTSPParams.html"><img src="right.png" width="16" height="16" border="0" alt="Next"></a></td>
+</tr></table>
+<div class="refentry">
+<a name="gst-rtsp-server-GstRTSPPermissions"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPPermissions.top_of_page"></a>GstRTSPPermissions</span></h2>
+<p>GstRTSPPermissions — Roles and associated permissions</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPPermissions.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-new" title="gst_rtsp_permissions_new ()">gst_rtsp_permissions_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-ref" title="gst_rtsp_permissions_ref ()">gst_rtsp_permissions_ref</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-unref" title="gst_rtsp_permissions_unref ()">gst_rtsp_permissions_unref</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role" title="gst_rtsp_permissions_add_role ()">gst_rtsp_permissions_add_role</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role-valist" title="gst_rtsp_permissions_add_role_valist ()">gst_rtsp_permissions_add_role_valist</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-remove-role" title="gst_rtsp_permissions_remove_role ()">gst_rtsp_permissions_remove_role</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-get-role" title="gst_rtsp_permissions_get_role ()">gst_rtsp_permissions_get_role</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-is-allowed" title="gst_rtsp_permissions_is_allowed ()">gst_rtsp_permissions_is_allowed</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPPermissions.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody><tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions">GstRTSPPermissions</a></td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPPermissions.description"></a><h2>Description</h2>
+<p>The <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> object contains an array of roles and associated
+permissions. The roles are represented with a string and the permissions with
+a generic <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="type">GstStructure</span></a>.</p>
+<p>The permissions are deliberately kept generic. The possible values of the
+roles and <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="type">GstStructure</span></a> keys and values are only determined by the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>
+object that performs the checks on the permissions and the current
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>.</p>
+<p>As a convenience function, <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-is-allowed" title="gst_rtsp_permissions_is_allowed ()"><code class="function">gst_rtsp_permissions_is_allowed()</code></a> can be used to
+check if the permissions contains a role that contains the boolean value
+<a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> for the the given key.</p>
+<p>Last reviewed on 2013-07-15 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPPermissions.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-permissions-new"></a><h3>gst_rtsp_permissions_new ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
+gst_rtsp_permissions_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Create a new empty Authorization permissions.</p>
+<div class="refsect3">
+<a name="id-1.2.18.6.2.5"></a><h4>Returns</h4>
+<p> a new empty authorization permissions. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-permissions-ref"></a><h3>gst_rtsp_permissions_ref ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
+gst_rtsp_permissions_ref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>);</pre>
+<p>Increase the refcount of this permissions.</p>
+<div class="refsect3">
+<a name="id-1.2.18.6.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>permissions</p></td>
+<td class="parameter_description"><p>The permissions to refcount</p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.18.6.3.6"></a><h4>Returns</h4>
+<p> <em class="parameter"><code>permissions</code></em>
+(for convenience when doing assignments). </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-permissions-unref"></a><h3>gst_rtsp_permissions_unref ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_permissions_unref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>);</pre>
+<p>Decrease the refcount of an permissions, freeing it if the refcount reaches 0.</p>
+<div class="refsect3">
+<a name="id-1.2.18.6.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>permissions</p></td>
+<td class="parameter_description"><p> the permissions to refcount. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-permissions-add-role"></a><h3>gst_rtsp_permissions_add_role ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_permissions_add_role (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *fieldname</code></em>,
+ <em class="parameter"><code>...</code></em>);</pre>
+<p>Add a new <em class="parameter"><code>role</code></em>
+ to <em class="parameter"><code>permissions</code></em>
+ with the given variables. The fields
+are the same layout as <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-new"><code class="function">gst_structure_new()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.18.6.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>permissions</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>role</p></td>
+<td class="parameter_description"><p>a role</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>fieldname</p></td>
+<td class="parameter_description"><p>the first field name</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>...</p></td>
+<td class="parameter_description"><p>additional arguments</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-permissions-add-role-valist"></a><h3>gst_rtsp_permissions_add_role_valist ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_permissions_add_role_valist (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *fieldname</code></em>,
+ <em class="parameter"><code><span class="type">va_list</span> var_args</code></em>);</pre>
+<p>Add a new <em class="parameter"><code>role</code></em>
+ to <em class="parameter"><code>permissions</code></em>
+ with the given variables. Structure fields
+are set according to the varargs in a manner similar to <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-new"><code class="function">gst_structure_new()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.18.6.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>permissions</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>role</p></td>
+<td class="parameter_description"><p>a role</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>fieldname</p></td>
+<td class="parameter_description"><p>the first field name</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>var_args</p></td>
+<td class="parameter_description"><p>additional fields to add</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-permissions-remove-role"></a><h3>gst_rtsp_permissions_remove_role ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_permissions_remove_role (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>);</pre>
+<p>Remove all permissions for <em class="parameter"><code>role</code></em>
+ in <em class="parameter"><code>permissions</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.18.6.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>permissions</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>role</p></td>
+<td class="parameter_description"><p>a role</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-permissions-get-role"></a><h3>gst_rtsp_permissions_get_role ()</h3>
+<pre class="programlisting">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
+gst_rtsp_permissions_get_role (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>);</pre>
+<p>Get all permissions for <em class="parameter"><code>role</code></em>
+ in <em class="parameter"><code>permissions</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.18.6.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>permissions</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>role</p></td>
+<td class="parameter_description"><p>a role</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.18.6.8.6"></a><h4>Returns</h4>
+<p> the structure with permissions for <em class="parameter"><code>role</code></em>
+. It
+remains valid for as long as <em class="parameter"><code>permissions</code></em>
+is valid. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-permissions-is-allowed"></a><h3>gst_rtsp_permissions_is_allowed ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_permissions_is_allowed (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *permission</code></em>);</pre>
+<p>Check if <em class="parameter"><code>role</code></em>
+ in <em class="parameter"><code>permissions</code></em>
+ is given permission for <em class="parameter"><code>permission</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.18.6.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>permissions</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>role</p></td>
+<td class="parameter_description"><p>a role</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>permission</p></td>
+<td class="parameter_description"><p>a permission</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.18.6.9.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>role</code></em>
+is allowed <em class="parameter"><code>permission</code></em>
+.</p>
+<p></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPPermissions.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPPermissions"></a><h3>struct GstRTSPPermissions</h3>
+<pre class="programlisting">struct GstRTSPPermissions {
+ GstMiniObject mini_object;
+};
+</pre>
+<p>The opaque permissions structure. It is used to define the permissions
+of objects in different roles.</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPPermissions.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>, <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
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+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#gst-rtsp-server-GstRTSPSdp.description" class="shortcut">Description</a></span>
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+</tr></table>
+<div class="refentry">
+<a name="gst-rtsp-server-GstRTSPSdp"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPSdp.top_of_page"></a>GstRTSPSdp</span></h2>
+<p>GstRTSPSdp — Make SDP messages</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSdp.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody><tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSdp.html#gst-rtsp-sdp-from-media" title="gst_rtsp_sdp_from_media ()">gst_rtsp_sdp_from_media</a> <span class="c_punctuation">()</span>
+</td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSdp.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody><tr>
+<td class="datatype_keyword"> </td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo">GstSDPInfo</a></td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSdp.description"></a><h2>Description</h2>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSdp.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-sdp-from-media"></a><h3>gst_rtsp_sdp_from_media ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_sdp_from_media (<em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstsdpmessage.html#GstSDPMessage"><span class="type">GstSDPMessage</span></a> *sdp</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo"><span class="type">GstSDPInfo</span></a> *info</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Add <em class="parameter"><code>media</code></em>
+ specific info to <em class="parameter"><code>sdp</code></em>
+. <em class="parameter"><code>info</code></em>
+ is used to configure the connection
+information in the SDP.</p>
+<div class="refsect3">
+<a name="id-1.2.13.6.2.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>sdp</p></td>
+<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstsdpmessage.html#GstSDPMessage"><span class="type">GstSDPMessage</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>info</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo"><span class="type">GstSDPInfo</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p> a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.13.6.2.6"></a><h4>Returns</h4>
+<p> TRUE on success.</p>
+<p></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSdp.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstSDPInfo"></a><h3>GstSDPInfo</h3>
+<pre class="programlisting">typedef struct {
+ gboolean is_ipv6;
+ const gchar *server_ip;
+} GstSDPInfo;
+</pre>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSdp.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
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+<title>GStreamer RTSP Server Reference Manual: GstRTSPSessionMedia</title>
+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
+<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
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+<link rel="prev" href="GstRTSPSession.html" title="GstRTSPSession">
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+<table class="navigation" id="top" width="100%" summary="Navigation header" cellpadding="2" cellspacing="5"><tr valign="middle">
+<td width="100%" align="left" class="shortcuts">
+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#gst-rtsp-server-GstRTSPSessionMedia.description" class="shortcut">Description</a></span>
+</td>
+<td><a accesskey="h" href="index.html"><img src="home.png" width="16" height="16" border="0" alt="Home"></a></td>
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+<td><a accesskey="n" href="gst-rtsp-server-GstRTSPStreamTransport.html"><img src="right.png" width="16" height="16" border="0" alt="Next"></a></td>
+</tr></table>
+<div class="refentry">
+<a name="gst-rtsp-server-GstRTSPSessionMedia"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPSessionMedia.top_of_page"></a>GstRTSPSessionMedia</span></h2>
+<p>GstRTSPSessionMedia — Media managed in a session</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSessionMedia.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-new" title="gst_rtsp_session_media_new ()">gst_rtsp_session_media_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-matches" title="gst_rtsp_session_media_matches ()">gst_rtsp_session_media_matches</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-media" title="gst_rtsp_session_media_get_media ()">gst_rtsp_session_media_get_media</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="returnvalue">GstClockTime</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-base-time" title="gst_rtsp_session_media_get_base_time ()">gst_rtsp_session_media_get_base_time</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtpinfo" title="gst_rtsp_session_media_get_rtpinfo ()">gst_rtsp_session_media_get_rtpinfo</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-state" title="gst_rtsp_session_media_set_state ()">gst_rtsp_session_media_set_state</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPState"><span class="returnvalue">GstRTSPState</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtsp-state" title="gst_rtsp_session_media_get_rtsp_state ()">gst_rtsp_session_media_get_rtsp_state</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-rtsp-state" title="gst_rtsp_session_media_set_rtsp_state ()">gst_rtsp_session_media_set_rtsp_state</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-transport" title="gst_rtsp_session_media_get_transport ()">gst_rtsp_session_media_get_transport</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-transport" title="gst_rtsp_session_media_set_transport ()">gst_rtsp_session_media_set_transport</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-alloc-channels" title="gst_rtsp_session_media_alloc_channels ()">gst_rtsp_session_media_alloc_channels</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSessionMedia.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia">GstRTSPSessionMedia</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMediaClass" title="struct GstRTSPSessionMediaClass">GstRTSPSessionMediaClass</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSessionMedia.description"></a><h2>Description</h2>
+<p>The <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> object manages a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> with a given path.</p>
+<p>With <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-transport" title="gst_rtsp_session_media_get_transport ()"><code class="function">gst_rtsp_session_media_get_transport()</code></a> and
+<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-transport" title="gst_rtsp_session_media_set_transport ()"><code class="function">gst_rtsp_session_media_set_transport()</code></a> the transports of a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> of
+the managed <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> can be retrieved and configured.</p>
+<p>Use <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-state" title="gst_rtsp_session_media_set_state ()"><code class="function">gst_rtsp_session_media_set_state()</code></a> to control the media state and
+transports.</p>
+<p>Last reviewed on 2013-07-16 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSessionMedia.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-new"></a><h3>gst_rtsp_session_media_new ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
+gst_rtsp_session_media_new (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
+<p>Create a new <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> that manages the streams
+in <em class="parameter"><code>media</code></em>
+ for <em class="parameter"><code>path</code></em>
+. <em class="parameter"><code>media</code></em>
+ should be prepared.</p>
+<p>Ownership is taken of <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.2.6"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>path</p></td>
+<td class="parameter_description"><p>the path</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p> the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.11.6.2.7"></a><h4>Returns</h4>
+<p> a new <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-matches"></a><h3>gst_rtsp_session_media_matches ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_session_media_matches (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> *matched</code></em>);</pre>
+<p>Check if the path of <em class="parameter"><code>media</code></em>
+ matches <em class="parameter"><code>path</code></em>
+. It <em class="parameter"><code>path</code></em>
+ matches, the amount of
+matched characters is returned in <em class="parameter"><code>matched</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>path</p></td>
+<td class="parameter_description"><p>a path</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>matched</p></td>
+<td class="parameter_description"><p> the amount of matched characters of <em class="parameter"><code>path</code></em>
+. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.11.6.3.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> when <em class="parameter"><code>path</code></em>
+matches the path of <em class="parameter"><code>media</code></em>
+.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-get-media"></a><h3>gst_rtsp_session_media_get_media ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
+gst_rtsp_session_media_get_media (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>);</pre>
+<p>Get the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> that was used when constructing <em class="parameter"><code>media</code></em>
+</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.11.6.4.6"></a><h4>Returns</h4>
+<p> the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> of <em class="parameter"><code>media</code></em>
+. Remains valid as long
+as <em class="parameter"><code>media</code></em>
+is valid. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-get-base-time"></a><h3>gst_rtsp_session_media_get_base_time ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="returnvalue">GstClockTime</span></a>
+gst_rtsp_session_media_get_base_time (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>);</pre>
+<p>Get the base_time of the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> in <em class="parameter"><code>media</code></em>
+</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.11.6.5.6"></a><h4>Returns</h4>
+<p> the base_time of the media.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-get-rtpinfo"></a><h3>gst_rtsp_session_media_get_rtpinfo ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_session_media_get_rtpinfo (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>);</pre>
+<p>Retrieve the RTP-Info header string for all streams in <em class="parameter"><code>media</code></em>
+
+with configured transports.</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.11.6.6.6"></a><h4>Returns</h4>
+<p> The RTP-Info as a string or
+<a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when no RTP-Info could be generated, <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-set-state"></a><h3>gst_rtsp_session_media_set_state ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_session_media_set_state (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GstState"><span class="type">GstState</span></a> state</code></em>);</pre>
+<p>Tell the media object <em class="parameter"><code>media</code></em>
+ to change to <em class="parameter"><code>state</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>state</p></td>
+<td class="parameter_description"><p>the new state</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.11.6.7.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-get-rtsp-state"></a><h3>gst_rtsp_session_media_get_rtsp_state ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPState"><span class="returnvalue">GstRTSPState</span></a>
+gst_rtsp_session_media_get_rtsp_state (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>);</pre>
+<p>Get the current RTSP state of <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.11.6.8.6"></a><h4>Returns</h4>
+<p> the current RTSP state of <em class="parameter"><code>media</code></em>
+.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-set-rtsp-state"></a><h3>gst_rtsp_session_media_set_rtsp_state ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_session_media_set_rtsp_state (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPState"><span class="type">GstRTSPState</span></a> state</code></em>);</pre>
+<p>Set the RTSP state of <em class="parameter"><code>media</code></em>
+ to <em class="parameter"><code>state</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>state</p></td>
+<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPState"><span class="type">GstRTSPState</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-get-transport"></a><h3>gst_rtsp_session_media_get_transport ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
+gst_rtsp_session_media_get_transport (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> idx</code></em>);</pre>
+<p>Get a previously created <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> for the stream at <em class="parameter"><code>idx</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>idx</p></td>
+<td class="parameter_description"><p>the stream index</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.11.6.10.6"></a><h4>Returns</h4>
+<p> a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> that is valid until the
+session of <em class="parameter"><code>media</code></em>
+is unreffed. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-set-transport"></a><h3>gst_rtsp_session_media_set_transport ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
+gst_rtsp_session_media_set_transport (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a> *tr</code></em>);</pre>
+<p>Configure the transport for <em class="parameter"><code>stream</code></em>
+ to <em class="parameter"><code>tr</code></em>
+ in <em class="parameter"><code>media</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.11.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>tr</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.11.6.11.6"></a><h4>Returns</h4>
+<p> the new or updated <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> for <em class="parameter"><code>stream</code></em>
+. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-session-media-alloc-channels"></a><h3>gst_rtsp_session_media_alloc_channels ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_session_media_alloc_channels (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPRange"><span class="type">GstRTSPRange</span></a> *range</code></em>);</pre>
+<p>Fill <em class="parameter"><code>range</code></em>
+ with the next available min and max channels for
+interleaved transport.</p>
+<div class="refsect3">
+<a name="id-1.2.11.6.12.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>media</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>range</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPRange"><span class="type">GstRTSPRange</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.11.6.12.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
+<p></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSessionMedia.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPSessionMedia"></a><h3>struct GstRTSPSessionMedia</h3>
+<pre class="programlisting">struct GstRTSPSessionMedia {
+ GObject parent;
+};
+</pre>
+<p>State of a client session regarding a specific media identified by path.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPSessionMediaClass"></a><h3>struct GstRTSPSessionMediaClass</h3>
+<pre class="programlisting">struct GstRTSPSessionMediaClass {
+ GObjectClass parent_class;
+};
+</pre>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPSessionMedia.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>, <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
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+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#gst-rtsp-server-GstRTSPStreamTransport.description" class="shortcut">Description</a></span>
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+</tr></table>
+<div class="refentry">
+<a name="gst-rtsp-server-GstRTSPStreamTransport"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPStreamTransport.top_of_page"></a>GstRTSPStreamTransport</span></h2>
+<p>GstRTSPStreamTransport — A media stream transport configuration</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPStreamTransport.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-new" title="gst_rtsp_stream_transport_new ()">gst_rtsp_stream_transport_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-stream" title="gst_rtsp_stream_transport_get_stream ()">gst_rtsp_stream_transport_get_stream</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="returnvalue">GstRTSPTransport</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-transport" title="gst_rtsp_stream_transport_get_transport ()">gst_rtsp_stream_transport_get_transport</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-transport" title="gst_rtsp_stream_transport_set_transport ()">gst_rtsp_stream_transport_set_transport</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="returnvalue">GstRTSPUrl</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-url" title="gst_rtsp_stream_transport_get_url ()">gst_rtsp_stream_transport_get_url</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-url" title="gst_rtsp_stream_transport_set_url ()">gst_rtsp_stream_transport_set_url</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-rtpinfo" title="gst_rtsp_stream_transport_get_rtpinfo ()">gst_rtsp_stream_transport_get_rtpinfo</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<span class="c_punctuation">(</span><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPSendFunc" title="GstRTSPSendFunc ()">*GstRTSPSendFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-callbacks" title="gst_rtsp_stream_transport_set_callbacks ()">gst_rtsp_stream_transport_set_callbacks</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<span class="c_punctuation">(</span><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPKeepAliveFunc" title="GstRTSPKeepAliveFunc ()">*GstRTSPKeepAliveFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-keepalive" title="gst_rtsp_stream_transport_set_keepalive ()">gst_rtsp_stream_transport_set_keepalive</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-keep-alive" title="gst_rtsp_stream_transport_keep_alive ()">gst_rtsp_stream_transport_keep_alive</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-active" title="gst_rtsp_stream_transport_set_active ()">gst_rtsp_stream_transport_set_active</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-timed-out" title="gst_rtsp_stream_transport_set_timed_out ()">gst_rtsp_stream_transport_set_timed_out</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-is-timed-out" title="gst_rtsp_stream_transport_is_timed_out ()">gst_rtsp_stream_transport_is_timed_out</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtcp" title="gst_rtsp_stream_transport_send_rtcp ()">gst_rtsp_stream_transport_send_rtcp</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtp" title="gst_rtsp_stream_transport_send_rtp ()">gst_rtsp_stream_transport_send_rtp</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPStreamTransport.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport">GstRTSPStreamTransport</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransportClass" title="struct GstRTSPStreamTransportClass">GstRTSPStreamTransportClass</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPStreamTransport.description"></a><h2>Description</h2>
+<p>The <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> configures the transport used by a
+<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>. It is usually manages by a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> object.</p>
+<p>With <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-callbacks" title="gst_rtsp_stream_transport_set_callbacks ()"><code class="function">gst_rtsp_stream_transport_set_callbacks()</code></a>, callbacks can be configured
+to handle the RTP and RTCP packets from the stream, for example when they
+need to be sent over TCP.</p>
+<p>With <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-active" title="gst_rtsp_stream_transport_set_active ()"><code class="function">gst_rtsp_stream_transport_set_active()</code></a> the transports are added and
+removed from the stream.</p>
+<p>A <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> will call <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-keep-alive" title="gst_rtsp_stream_transport_keep_alive ()"><code class="function">gst_rtsp_stream_transport_keep_alive()</code></a> when RTCP
+is received from the client. It will also call
+<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-timed-out" title="gst_rtsp_stream_transport_set_timed_out ()"><code class="function">gst_rtsp_stream_transport_set_timed_out()</code></a> when a receiver has timed out.</p>
+<p>Last reviewed on 2013-07-16 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPStreamTransport.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-new"></a><h3>gst_rtsp_stream_transport_new ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
+gst_rtsp_stream_transport_new (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a> *tr</code></em>);</pre>
+<p>Create a new <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> that can be used to manage
+<em class="parameter"><code>stream</code></em>
+ with transport <em class="parameter"><code>tr</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.2.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>stream</p></td>
+<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>tr</p></td>
+<td class="parameter_description"><p> a GstRTSPTransport. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.12.6.2.6"></a><h4>Returns</h4>
+<p> a new <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-get-stream"></a><h3>gst_rtsp_stream_transport_get_stream ()</h3>
+<pre class="programlisting"><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
+gst_rtsp_stream_transport_get_stream (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
+<p>Get the <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> used when constructing <em class="parameter"><code>trans</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.12.6.3.6"></a><h4>Returns</h4>
+<p> the stream used when constructing <em class="parameter"><code>trans</code></em>
+. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-get-transport"></a><h3>gst_rtsp_stream_transport_get_transport ()</h3>
+<pre class="programlisting">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="returnvalue">GstRTSPTransport</span></a> *
+gst_rtsp_stream_transport_get_transport
+ (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
+<p>Get the transport configured in <em class="parameter"><code>trans</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.12.6.4.6"></a><h4>Returns</h4>
+<p> the transport configured in <em class="parameter"><code>trans</code></em>
+. It remains
+valid for as long as <em class="parameter"><code>trans</code></em>
+is valid. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-set-transport"></a><h3>gst_rtsp_stream_transport_set_transport ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_transport_set_transport
+ (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a> *tr</code></em>);</pre>
+<p>Set <em class="parameter"><code>tr</code></em>
+ as the client transport. This function takes ownership of the
+passed <em class="parameter"><code>tr</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>tr</p></td>
+<td class="parameter_description"><p> a client <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-get-url"></a><h3>gst_rtsp_stream_transport_get_url ()</h3>
+<pre class="programlisting">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="returnvalue">GstRTSPUrl</span></a> *
+gst_rtsp_stream_transport_get_url (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
+<p>Get the url configured in <em class="parameter"><code>trans</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.12.6.6.6"></a><h4>Returns</h4>
+<p> the url configured in <em class="parameter"><code>trans</code></em>
+. It remains
+valid for as long as <em class="parameter"><code>trans</code></em>
+is valid. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-set-url"></a><h3>gst_rtsp_stream_transport_set_url ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_transport_set_url (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
+ <em class="parameter"><code>const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a> *url</code></em>);</pre>
+<p>Set <em class="parameter"><code>url</code></em>
+ as the client url.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>url</p></td>
+<td class="parameter_description"><p> a client <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-get-rtpinfo"></a><h3>gst_rtsp_stream_transport_get_rtpinfo ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_stream_transport_get_rtpinfo (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a> start_time</code></em>);</pre>
+<p>Get the RTP-Info string for <em class="parameter"><code>trans</code></em>
+ and <em class="parameter"><code>start_time</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>start_time</p></td>
+<td class="parameter_description"><p>a star time</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.12.6.8.6"></a><h4>Returns</h4>
+<p> the RTPInfo string for <em class="parameter"><code>trans</code></em>
+and <em class="parameter"><code>start_time</code></em>
+or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when the RTP-Info could not be
+determined. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPSendFunc"></a><h3>GstRTSPSendFunc ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+<span class="c_punctuation">(</span>*GstRTSPSendFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a> *buffer</code></em>,
+ <em class="parameter"><code><span class="type">guint8</span> channel</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>Function registered with <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-callbacks" title="gst_rtsp_stream_transport_set_callbacks ()"><code class="function">gst_rtsp_stream_transport_set_callbacks()</code></a> and
+called when <em class="parameter"><code>buffer</code></em>
+ must be sent on <em class="parameter"><code>channel</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>buffer</p></td>
+<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>channel</p></td>
+<td class="parameter_description"><p>a channel</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.12.6.9.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-set-callbacks"></a><h3>gst_rtsp_stream_transport_set_callbacks ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_transport_set_callbacks
+ (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPSendFunc" title="GstRTSPSendFunc ()"><span class="type">GstRTSPSendFunc</span></a> send_rtp</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPSendFunc" title="GstRTSPSendFunc ()"><span class="type">GstRTSPSendFunc</span></a> send_rtcp</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Datasets.html#GDestroyNotify"><span class="type">GDestroyNotify</span></a> notify</code></em>);</pre>
+<p>Install callbacks that will be called when data for a stream should be sent
+to a client. This is usually used when sending RTP/RTCP over TCP.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>send_rtp</p></td>
+<td class="parameter_description"><p> a callback called when RTP should be sent. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid until the GDestroyNotify argument is called."><span class="acronym">scope notified</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>send_rtcp</p></td>
+<td class="parameter_description"><p> a callback called when RTCP should be sent. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid until the GDestroyNotify argument is called."><span class="acronym">scope notified</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p> user data passed to callbacks. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>notify</p></td>
+<td class="parameter_description"><p> called with the user_data when no longer needed. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPKeepAliveFunc"></a><h3>GstRTSPKeepAliveFunc ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+<span class="c_punctuation">(</span>*GstRTSPKeepAliveFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
+<p>Function registered with <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-keepalive" title="gst_rtsp_stream_transport_set_keepalive ()"><code class="function">gst_rtsp_stream_transport_set_keepalive()</code></a> and called
+when the stream is active.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.11.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p>user data</p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-set-keepalive"></a><h3>gst_rtsp_stream_transport_set_keepalive ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_transport_set_keepalive
+ (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPKeepAliveFunc" title="GstRTSPKeepAliveFunc ()"><span class="type">GstRTSPKeepAliveFunc</span></a> keep_alive</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Datasets.html#GDestroyNotify"><span class="type">GDestroyNotify</span></a> notify</code></em>);</pre>
+<p>Install callbacks that will be called when RTCP packets are received from the
+receiver of <em class="parameter"><code>trans</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.12.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>keep_alive</p></td>
+<td class="parameter_description"><p> a callback called when the receiver is active. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid until the GDestroyNotify argument is called."><span class="acronym">scope notified</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>user_data</p></td>
+<td class="parameter_description"><p> user data passed to callback. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
+</tr>
+<tr>
+<td class="parameter_name"><p>notify</p></td>
+<td class="parameter_description"><p> called with the user_data when no longer needed. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-keep-alive"></a><h3>gst_rtsp_stream_transport_keep_alive ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_transport_keep_alive (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
+<p>Signal the installed keep_alive callback for <em class="parameter"><code>trans</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.13.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-set-active"></a><h3>gst_rtsp_stream_transport_set_active ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_transport_set_active (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> active</code></em>);</pre>
+<p>Activate or deactivate datatransfer configured in <em class="parameter"><code>trans</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.14.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>active</p></td>
+<td class="parameter_description"><p>new state of <em class="parameter"><code>trans</code></em>
+</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.12.6.14.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> when the state was changed.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-set-timed-out"></a><h3>gst_rtsp_stream_transport_set_timed_out ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_stream_transport_set_timed_out
+ (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> timedout</code></em>);</pre>
+<p>Set the timed out state of <em class="parameter"><code>trans</code></em>
+ to <em class="parameter"><code>timedout</code></em>
+</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.15.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>timedout</p></td>
+<td class="parameter_description"><p>timed out value</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-is-timed-out"></a><h3>gst_rtsp_stream_transport_is_timed_out ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_transport_is_timed_out
+ (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
+<p>Check if <em class="parameter"><code>trans</code></em>
+ is timed out.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.16.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.12.6.16.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>trans</code></em>
+timed out.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-send-rtcp"></a><h3>gst_rtsp_stream_transport_send_rtcp ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_transport_send_rtcp (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a> *buffer</code></em>);</pre>
+<p>Send <em class="parameter"><code>buffer</code></em>
+ to the installed RTCP callback for <em class="parameter"><code>trans</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.17.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>buffer</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.12.6.17.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-stream-transport-send-rtp"></a><h3>gst_rtsp_stream_transport_send_rtp ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_stream_transport_send_rtp (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
+ <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a> *buffer</code></em>);</pre>
+<p>Send <em class="parameter"><code>buffer</code></em>
+ to the installed RTP callback for <em class="parameter"><code>trans</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.12.6.18.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>trans</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>buffer</p></td>
+<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.12.6.18.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success</p>
+<p></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPStreamTransport.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPStreamTransport"></a><h3>struct GstRTSPStreamTransport</h3>
+<pre class="programlisting">struct GstRTSPStreamTransport {
+ GObject parent;
+};
+</pre>
+<p>A Transport description for a stream</p>
+<div class="refsect3">
+<a name="id-1.2.12.7.2.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody><tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject"><span class="type">GObject</span></a> <em class="structfield"><code><a name="GstRTSPStreamTransport.parent"></a>parent</code></em>;</p></td>
+<td class="struct_member_description"><p>parent instance</p></td>
+<td class="struct_member_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPStreamTransportClass"></a><h3>struct GstRTSPStreamTransportClass</h3>
+<pre class="programlisting">struct GstRTSPStreamTransportClass {
+ GObjectClass parent_class;
+};
+</pre>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPStreamTransport.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
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+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
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+<td><a accesskey="n" href="GstRTSPAuth.html"><img src="right.png" width="16" height="16" border="0" alt="Next"></a></td>
+</tr></table>
+<div class="refentry">
+<a name="gst-rtsp-server-GstRTSPThreadPool"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPThreadPool.top_of_page"></a>GstRTSPThreadPool</span></h2>
+<p>GstRTSPThreadPool — A pool of threads</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPThreadPool.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-new" title="gst_rtsp_thread_new ()">gst_rtsp_thread_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-ref" title="gst_rtsp_thread_ref ()">gst_rtsp_thread_ref</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-unref" title="gst_rtsp_thread_unref ()">gst_rtsp_thread_unref</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-reuse" title="gst_rtsp_thread_reuse ()">gst_rtsp_thread_reuse</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-stop" title="gst_rtsp_thread_stop ()">gst_rtsp_thread_stop</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-new" title="gst_rtsp_thread_pool_new ()">gst_rtsp_thread_pool_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-max-threads" title="gst_rtsp_thread_pool_get_max_threads ()">gst_rtsp_thread_pool_get_max_threads</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-set-max-threads" title="gst_rtsp_thread_pool_set_max_threads ()">gst_rtsp_thread_pool_set_max_threads</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-thread" title="gst_rtsp_thread_pool_get_thread ()">gst_rtsp_thread_pool_get_thread</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-cleanup" title="gst_rtsp_thread_pool_cleanup ()">gst_rtsp_thread_pool_cleanup</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPThreadPool.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody>
+<tr>
+<td class="datatype_keyword">enum</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType">GstRTSPThreadType</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread">GstRTSPThread</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool">GstRTSPThreadPool</a></td>
+</tr>
+<tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPoolClass" title="struct GstRTSPThreadPoolClass">GstRTSPThreadPoolClass</a></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPThreadPool.description"></a><h2>Description</h2>
+<p>A <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> manages reusable threads for various server tasks.
+Currently the defined thread types can be found in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType"><span class="type">GstRTSPThreadType</span></a>.</p>
+<p>Threads of type <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GST-RTSP-THREAD-TYPE-CLIENT:CAPS"><span class="type">GST_RTSP_THREAD_TYPE_CLIENT</span></a> are used to handle requests from
+a connected client. With <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-max-threads" title="gst_rtsp_thread_pool_get_max_threads ()"><code class="function">gst_rtsp_thread_pool_get_max_threads()</code></a> a maximum
+number of threads can be set after which the pool will start to reuse the
+same thread for multiple clients.</p>
+<p>Threads of type <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GST-RTSP-THREAD-TYPE-MEDIA:CAPS"><span class="type">GST_RTSP_THREAD_TYPE_MEDIA</span></a> will be used to perform the state
+changes of the media pipelines and handle its bus messages.</p>
+<p>gst_rtsp_thread_pool_get_thread() can be used to create a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a>
+object of the right type. The thread object contains a mainloop and context
+that run in a seperate thread and can be used to attached sources to.</p>
+<p>gst_rtsp_thread_reuse() can be used to reuse a thread for multiple purposes.
+If all <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-reuse" title="gst_rtsp_thread_reuse ()"><code class="function">gst_rtsp_thread_reuse()</code></a> calls are matched with a
+<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-stop" title="gst_rtsp_thread_stop ()"><code class="function">gst_rtsp_thread_stop()</code></a> call, the mainloop will be quit and the thread will
+stop.</p>
+<p>To configure the threads, a subclass of this object should be made and the
+virtual methods should be overriden to implement the desired functionality.</p>
+<p>Last reviewed on 2013-07-11 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPThreadPool.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-thread-new"></a><h3>gst_rtsp_thread_new ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
+gst_rtsp_thread_new (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType"><span class="type">GstRTSPThreadType</span></a> type</code></em>);</pre>
+<p>Create a new thread object that can run a mainloop.</p>
+<div class="refsect3">
+<a name="id-1.2.15.6.2.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>type</p></td>
+<td class="parameter_description"><p>the thread type</p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.15.6.2.6"></a><h4>Returns</h4>
+<p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-thread-ref"></a><h3>gst_rtsp_thread_ref ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
+gst_rtsp_thread_ref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> *thread</code></em>);</pre>
+<p>Increase the refcount of this thread.</p>
+<div class="refsect3">
+<a name="id-1.2.15.6.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>thread</p></td>
+<td class="parameter_description"><p>The thread to refcount</p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.15.6.3.6"></a><h4>Returns</h4>
+<p> <em class="parameter"><code>thread</code></em>
+(for convenience when doing assignments). </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-thread-unref"></a><h3>gst_rtsp_thread_unref ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_thread_unref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *thread</code></em>);</pre>
+<p>Decrease the refcount of an thread, freeing it if the refcount reaches 0.</p>
+<div class="refsect3">
+<a name="id-1.2.15.6.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>thread</p></td>
+<td class="parameter_description"><p> the thread to refcount. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-thread-reuse"></a><h3>gst_rtsp_thread_reuse ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_thread_reuse (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> *thread</code></em>);</pre>
+<p>Reuse the mainloop of <em class="parameter"><code>thread</code></em>
+</p>
+<div class="refsect3">
+<a name="id-1.2.15.6.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>thread</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.15.6.5.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the mainloop could be reused</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-thread-stop"></a><h3>gst_rtsp_thread_stop ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_thread_stop (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> *thread</code></em>);</pre>
+<p>Stop and unref <em class="parameter"><code>thread</code></em>
+. When no threads are using the mainloop, the thread
+will be stopped and the final ref to <em class="parameter"><code>thread</code></em>
+ will be released.</p>
+<div class="refsect3">
+<a name="id-1.2.15.6.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>thread</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-thread-pool-new"></a><h3>gst_rtsp_thread_pool_new ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
+gst_rtsp_thread_pool_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Create a new <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> instance.</p>
+<div class="refsect3">
+<a name="id-1.2.15.6.7.5"></a><h4>Returns</h4>
+<p> a new <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a>. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-thread-pool-get-max-threads"></a><h3>gst_rtsp_thread_pool_get_max_threads ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
+gst_rtsp_thread_pool_get_max_threads (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> *pool</code></em>);</pre>
+<p>Get the maximum number of threads used for client connections.
+See <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-set-max-threads" title="gst_rtsp_thread_pool_set_max_threads ()"><code class="function">gst_rtsp_thread_pool_set_max_threads()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.15.6.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.15.6.8.6"></a><h4>Returns</h4>
+<p> the maximum number of threads.</p>
+<p></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-thread-pool-set-max-threads"></a><h3>gst_rtsp_thread_pool_set_max_threads ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_thread_pool_set_max_threads (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> *pool</code></em>,
+ <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> max_threads</code></em>);</pre>
+<p>Set the maximum threads used by the pool to handle client requests.
+A value of 0 will use the pool mainloop, a value of -1 will use an
+unlimited number of threads.</p>
+<div class="refsect3">
+<a name="id-1.2.15.6.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>max_threads</p></td>
+<td class="parameter_description"><p>maximum threads</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-thread-pool-get-thread"></a><h3>gst_rtsp_thread_pool_get_thread ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
+gst_rtsp_thread_pool_get_thread (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> *pool</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType"><span class="type">GstRTSPThreadType</span></a> type</code></em>,
+ <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *ctx</code></em>);</pre>
+<p>Get a new <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> for <em class="parameter"><code>type</code></em>
+ and <em class="parameter"><code>ctx</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.15.6.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>pool</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>type</p></td>
+<td class="parameter_description"><p>the <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType"><span class="type">GstRTSPThreadType</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>ctx</p></td>
+<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a>. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.15.6.10.6"></a><h4>Returns</h4>
+<p> a new <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-stop" title="gst_rtsp_thread_stop ()"><code class="function">gst_rtsp_thread_stop()</code></a> after usage. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-thread-pool-cleanup"></a><h3>gst_rtsp_thread_pool_cleanup ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_thread_pool_cleanup (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Wait for all tasks to be stopped and free all allocated resources. This is
+mainly used in test suites to ensure proper cleanup of internal data
+structures.</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPThreadPool.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPThreadType"></a><h3>enum GstRTSPThreadType</h3>
+<p>Different thread types</p>
+<div class="refsect3">
+<a name="id-1.2.15.7.2.4"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="enum_members_name">
+<col class="enum_members_description">
+<col width="200px" class="enum_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-THREAD-TYPE-CLIENT:CAPS"></a>GST_RTSP_THREAD_TYPE_CLIENT</p></td>
+<td class="enum_member_description">
+<p>a thread to handle the client communication</p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+<tr>
+<td class="enum_member_name"><p><a name="GST-RTSP-THREAD-TYPE-MEDIA:CAPS"></a>GST_RTSP_THREAD_TYPE_MEDIA</p></td>
+<td class="enum_member_description">
+<p>a thread to handle media </p>
+</td>
+<td class="enum_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPThread"></a><h3>struct GstRTSPThread</h3>
+<pre class="programlisting">struct GstRTSPThread {
+ GstMiniObject mini_object;
+
+ GstRTSPThreadType type;
+ GMainContext *context;
+ GMainLoop *loop;
+};
+</pre>
+<p>Structure holding info about a mainloop running in a thread</p>
+<div class="refsect3">
+<a name="id-1.2.15.7.3.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstMiniObject.html#GstMiniObject"><span class="type">GstMiniObject</span></a> <em class="structfield"><code><a name="GstRTSPThread.mini-object"></a>mini_object</code></em>;</p></td>
+<td class="struct_member_description"><p>parent <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstMiniObject.html#GstMiniObject"><span class="type">GstMiniObject</span></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType"><span class="type">GstRTSPThreadType</span></a> <em class="structfield"><code><a name="GstRTSPThread.type"></a>type</code></em>;</p></td>
+<td class="struct_member_description"><p>the thread type</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a> *<em class="structfield"><code><a name="GstRTSPThread.context"></a>context</code></em>;</p></td>
+<td class="struct_member_description"><p>a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainLoop"><span class="type">GMainLoop</span></a> *<em class="structfield"><code><a name="GstRTSPThread.loop"></a>loop</code></em>;</p></td>
+<td class="struct_member_description"><p>a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainLoop"><span class="type">GMainLoop</span></a></p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPThreadPool"></a><h3>struct GstRTSPThreadPool</h3>
+<pre class="programlisting">struct GstRTSPThreadPool {
+ GObject parent;
+};
+</pre>
+<p>The thread pool structure.</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTSPThreadPoolClass"></a><h3>struct GstRTSPThreadPoolClass</h3>
+<pre class="programlisting">struct GstRTSPThreadPoolClass {
+ GObjectClass parent_class;
+
+ GThreadPool *pool;
+
+ GstRTSPThread * (*get_thread) (GstRTSPThreadPool *pool,
+ GstRTSPThreadType type,
+ GstRTSPContext *ctx);
+ void (*configure_thread) (GstRTSPThreadPool *pool,
+ GstRTSPThread * thread,
+ GstRTSPContext *ctx);
+
+ void (*thread_enter) (GstRTSPThreadPool *pool,
+ GstRTSPThread *thread);
+ void (*thread_leave) (GstRTSPThreadPool *pool,
+ GstRTSPThread *thread);
+};
+</pre>
+<p>Class for managing threads.</p>
+<div class="refsect3">
+<a name="id-1.2.15.7.5.5"></a><h4>Members</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="300px" class="struct_members_name">
+<col class="struct_members_description">
+<col width="200px" class="struct_members_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPThreadPoolClass.parent-class"></a>parent_class</code></em>;</p></td>
+<td> </td>
+<td> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><a href="https://developer.gnome.org/glib/unstable/glib-Thread-Pools.html#GThreadPool"><span class="type">GThreadPool</span></a> *<em class="structfield"><code><a name="GstRTSPThreadPoolClass.pool"></a>pool</code></em>;</p></td>
+<td class="struct_member_description"><p>a <a href="https://developer.gnome.org/glib/unstable/glib-Thread-Pools.html#GThreadPool"><span class="type">GThreadPool</span></a> used internally</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPThreadPoolClass.get-thread"></a>get_thread</code></em> ()</p></td>
+<td class="struct_member_description"><p>this function should make or reuse an existing thread that runs
+a mainloop.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPThreadPoolClass.configure-thread"></a>configure_thread</code></em> ()</p></td>
+<td class="struct_member_description"><p>configure a thread object. this vmethod is called when
+a new thread has been created and should be configured.</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPThreadPoolClass.thread-enter"></a>thread_enter</code></em> ()</p></td>
+<td class="struct_member_description"><p>called from the thread when it is entered</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+<tr>
+<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPThreadPoolClass.thread-leave"></a>thread_leave</code></em> ()</p></td>
+<td class="struct_member_description"><p>called from the thread when it is left</p></td>
+<td class="struct_member_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPThreadPool.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>, <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p>
+</div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
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+<title>GStreamer RTSP Server Reference Manual: GstRTSPToken</title>
+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
+<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
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+<td width="100%" align="left" class="shortcuts">
+<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
+ <a href="#gst-rtsp-server-GstRTSPToken.description" class="shortcut">Description</a></span>
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+</tr></table>
+<div class="refentry">
+<a name="gst-rtsp-server-GstRTSPToken"></a><div class="titlepage"></div>
+<div class="refnamediv"><table width="100%"><tr>
+<td valign="top">
+<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPToken.top_of_page"></a>GstRTSPToken</span></h2>
+<p>GstRTSPToken — Roles and permissions for a client</p>
+</td>
+<td class="gallery_image" valign="top" align="right"></td>
+</tr></table></div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPToken.functions"></a><h2>Functions</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="functions_return">
+<col class="functions_name">
+</colgroup>
+<tbody>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-empty" title="gst_rtsp_token_new_empty ()">gst_rtsp_token_new_empty</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new" title="gst_rtsp_token_new ()">gst_rtsp_token_new</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-valist" title="gst_rtsp_token_new_valist ()">gst_rtsp_token_new_valist</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-ref" title="gst_rtsp_token_ref ()">gst_rtsp_token_ref</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<span class="returnvalue">void</span>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-unref" title="gst_rtsp_token_unref ()">gst_rtsp_token_unref</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-structure" title="gst_rtsp_token_get_structure ()">gst_rtsp_token_get_structure</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-writable-structure" title="gst_rtsp_token_writable_structure ()">gst_rtsp_token_writable_structure</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-string" title="gst_rtsp_token_get_string ()">gst_rtsp_token_get_string</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+<tr>
+<td class="function_type">
+<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+</td>
+<td class="function_name">
+<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-is-allowed" title="gst_rtsp_token_is_allowed ()">gst_rtsp_token_is_allowed</a> <span class="c_punctuation">()</span>
+</td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPToken.other"></a><h2>Types and Values</h2>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="name">
+<col class="description">
+</colgroup>
+<tbody><tr>
+<td class="datatype_keyword">struct</td>
+<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken">GstRTSPToken</a></td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPToken.description"></a><h2>Description</h2>
+<p>A <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> contains the permissions and roles of the user
+performing the current request. A token is usually created when a user is
+authenticated by the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> object and is then placed as the current
+token for the current request.</p>
+<p><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> can use the token and its contents to check authorization for
+various operations by comparing the token to the <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> of the
+object.</p>
+<p>The accepted values of the token are entirely defined by the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>
+object that implements the security policy.</p>
+<p>Last reviewed on 2013-07-15 (1.0.0)</p>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPToken.functions_details"></a><h2>Functions</h2>
+<div class="refsect2">
+<a name="gst-rtsp-token-new-empty"></a><h3>gst_rtsp_token_new_empty ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
+gst_rtsp_token_new_empty (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
+<p>Create a new empty Authorization token.</p>
+<div class="refsect3">
+<a name="id-1.2.17.6.2.5"></a><h4>Returns</h4>
+<p> a new empty authorization token. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-token-new"></a><h3>gst_rtsp_token_new ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
+gst_rtsp_token_new (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *firstfield</code></em>,
+ <em class="parameter"><code>...</code></em>);</pre>
+<p>Create a new Authorization token with the given fieldnames and values.
+Arguments are given similar to <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-new"><code class="function">gst_structure_new()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.17.6.3.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>firstfield</p></td>
+<td class="parameter_description"><p>the first fieldname</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>...</p></td>
+<td class="parameter_description"><p>additional arguments</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.17.6.3.6"></a><h4>Returns</h4>
+<p> a new authorization token. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-token-new-valist"></a><h3>gst_rtsp_token_new_valist ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
+gst_rtsp_token_new_valist (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *firstfield</code></em>,
+ <em class="parameter"><code><span class="type">va_list</span> var_args</code></em>);</pre>
+<p>Create a new Authorization token with the given fieldnames and values.
+Arguments are given similar to <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-new-valist"><code class="function">gst_structure_new_valist()</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.17.6.4.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>firstfield</p></td>
+<td class="parameter_description"><p>the first fieldname</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>var_args</p></td>
+<td class="parameter_description"><p>additional arguments</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.17.6.4.6"></a><h4>Returns</h4>
+<p> a new authorization token. </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-token-ref"></a><h3>gst_rtsp_token_ref ()</h3>
+<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
+gst_rtsp_token_ref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
+<p>Increase the refcount of this token.</p>
+<div class="refsect3">
+<a name="id-1.2.17.6.5.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>token</p></td>
+<td class="parameter_description"><p>The token to refcount</p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.17.6.5.6"></a><h4>Returns</h4>
+<p> <em class="parameter"><code>token</code></em>
+(for convenience when doing assignments). </p>
+<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-token-unref"></a><h3>gst_rtsp_token_unref ()</h3>
+<pre class="programlisting"><span class="returnvalue">void</span>
+gst_rtsp_token_unref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
+<p>Decrease the refcount of an token, freeing it if the refcount reaches 0.</p>
+<div class="refsect3">
+<a name="id-1.2.17.6.6.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>token</p></td>
+<td class="parameter_description"><p> the token to refcount. </p></td>
+<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
+</tr></tbody>
+</table></div>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-token-get-structure"></a><h3>gst_rtsp_token_get_structure ()</h3>
+<pre class="programlisting">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
+gst_rtsp_token_get_structure (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
+<p>Access the structure of the token.</p>
+<div class="refsect3">
+<a name="id-1.2.17.6.7.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>token</p></td>
+<td class="parameter_description"><p>The <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>.</p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.17.6.7.6"></a><h4>Returns</h4>
+<p> The structure of the token. The structure is still
+owned by the token, which means that you should not free it and that the
+pointer becomes invalid when you free the token.</p>
+<p>MT safe. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-token-writable-structure"></a><h3>gst_rtsp_token_writable_structure ()</h3>
+<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
+gst_rtsp_token_writable_structure (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
+<p>Get a writable version of the structure.</p>
+<div class="refsect3">
+<a name="id-1.2.17.6.8.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody><tr>
+<td class="parameter_name"><p>token</p></td>
+<td class="parameter_description"><p>The <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>.</p></td>
+<td class="parameter_annotations"> </td>
+</tr></tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.17.6.8.6"></a><h4>Returns</h4>
+<p> The structure of the token. The structure is still
+owned by the token, which means that you should not free it and that the
+pointer becomes invalid when you free the token. This function checks if
+<em class="parameter"><code>token</code></em>
+is writable and will never return <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>.</p>
+<p>MT safe. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-token-get-string"></a><h3>gst_rtsp_token_get_string ()</h3>
+<pre class="programlisting">const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
+gst_rtsp_token_get_string (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *field</code></em>);</pre>
+<p>Get the string value of <em class="parameter"><code>field</code></em>
+ in <em class="parameter"><code>token</code></em>
+.</p>
+<div class="refsect3">
+<a name="id-1.2.17.6.9.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>token</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>field</p></td>
+<td class="parameter_description"><p>a field name</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.17.6.9.6"></a><h4>Returns</h4>
+<p> the string value of <em class="parameter"><code>field</code></em>
+in
+<em class="parameter"><code>token</code></em>
+or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when <em class="parameter"><code>field</code></em>
+is not defined in <em class="parameter"><code>token</code></em>
+. The string
+becomes invalid when you free <em class="parameter"><code>token</code></em>
+. </p>
+<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
+</div>
+</div>
+<hr>
+<div class="refsect2">
+<a name="gst-rtsp-token-is-allowed"></a><h3>gst_rtsp_token_is_allowed ()</h3>
+<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
+gst_rtsp_token_is_allowed (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>,
+ <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *field</code></em>);</pre>
+<p>Check if <em class="parameter"><code>token</code></em>
+ has a boolean <em class="parameter"><code>field</code></em>
+ and if it is set to <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a>.</p>
+<div class="refsect3">
+<a name="id-1.2.17.6.10.5"></a><h4>Parameters</h4>
+<div class="informaltable"><table width="100%" border="0">
+<colgroup>
+<col width="150px" class="parameters_name">
+<col class="parameters_description">
+<col width="200px" class="parameters_annotations">
+</colgroup>
+<tbody>
+<tr>
+<td class="parameter_name"><p>token</p></td>
+<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a></p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+<tr>
+<td class="parameter_name"><p>field</p></td>
+<td class="parameter_description"><p>a field name</p></td>
+<td class="parameter_annotations"> </td>
+</tr>
+</tbody>
+</table></div>
+</div>
+<div class="refsect3">
+<a name="id-1.2.17.6.10.6"></a><h4>Returns</h4>
+<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>token</code></em>
+has a boolean field named <em class="parameter"><code>field</code></em>
+set to <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a>.</p>
+<p></p>
+</div>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPToken.other_details"></a><h2>Types and Values</h2>
+<div class="refsect2">
+<a name="GstRTSPToken"></a><h3>struct GstRTSPToken</h3>
+<pre class="programlisting">struct GstRTSPToken {
+ GstMiniObject mini_object;
+};
+</pre>
+<p>An opaque object used for checking authorisations.
+It is generated after successful authentication.</p>
+</div>
+</div>
+<div class="refsect1">
+<a name="gst-rtsp-server-GstRTSPToken.see-also"></a><h2>See Also</h2>
+<p><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a>, <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p>
+</div>
+</div>
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+<hr>
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+<div><p class="releaseinfo">
+ for GStreamer RTSP Server 1.4.5
+ </p></div>
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+<hr>
+</div>
+<div class="toc"><dl class="toc">
+<dt><span class="chapter"><a href="ch01.html"></a></span></dt>
+<dd><dl>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPServer.html">GstRTSPServer</a></span><span class="refpurpose"> — The main server object</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPClient.html">GstRTSPClient</a></span><span class="refpurpose"> — A client connection state</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPContext.html">GstRTSPContext</a></span><span class="refpurpose"> — A client request context</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPMountPoints.html">GstRTSPMountPoints</a></span><span class="refpurpose"> — Map a path to media</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPMediaFactory.html">GstRTSPMediaFactory</a></span><span class="refpurpose"> — A factory for media pipelines</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPMediaFactoryURI.html">GstRTSPMediaFactoryURI</a></span><span class="refpurpose"> — A factory for URI sources</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPMedia.html">GstRTSPMedia</a></span><span class="refpurpose"> — The media pipeline</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPStream.html">GstRTSPStream</a></span><span class="refpurpose"> — A media stream</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPSessionPool.html">GstRTSPSessionPool</a></span><span class="refpurpose"> — An object for managing sessions</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPSession.html">GstRTSPSession</a></span><span class="refpurpose"> — An object to manage media</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPSessionMedia.html">GstRTSPSessionMedia</a></span><span class="refpurpose"> — Media managed in a session</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPStreamTransport.html">GstRTSPStreamTransport</a></span><span class="refpurpose"> — A media stream transport configuration</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPSdp.html">GstRTSPSdp</a></span><span class="refpurpose"> — Make SDP messages</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPAddressPool.html">GstRTSPAddressPool</a></span><span class="refpurpose"> — A pool of network addresses</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPThreadPool.html">GstRTSPThreadPool</a></span><span class="refpurpose"> — A pool of threads</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="GstRTSPAuth.html">GstRTSPAuth</a></span><span class="refpurpose"> — Authentication and authorization</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPToken.html">GstRTSPToken</a></span><span class="refpurpose"> — Roles and permissions for a client</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPPermissions.html">GstRTSPPermissions</a></span><span class="refpurpose"> — Roles and associated permissions</span>
+</dt>
+<dt>
+<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPParams.html">GstRTSPParams</a></span><span class="refpurpose"> — Param get and set implementation</span>
+</dt>
+</dl></dd>
+<dt><span class="chapter"><a href="rtsp-server-hierarchy.html">Object Hierarchy</a></span></dt>
+<dt><span class="index"><a href="api-index-full.html">API Index</a></span></dt>
+<dt><span class="glossary"><a href="annotation-glossary.html">Annotation Glossary</a></span></dt>
+</dl></div>
+</div>
+<div class="footer">
+<hr>
+ Generated by GTK-Doc V1.21</div>
+</body>
+</html>
\ No newline at end of file
--- /dev/null
+<ANCHOR id="GstRTSPServer" href="gst-rtsp-server-1.0/GstRTSPServer.html">
+<ANCHOR id="GstRTSPServer.functions" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.functions">
+<ANCHOR id="GstRTSPServer.properties" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.properties">
+<ANCHOR id="GstRTSPServer.signals" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.signals">
+<ANCHOR id="GstRTSPServer.other" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.other">
+<ANCHOR id="GstRTSPServer.object-hierarchy" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.object-hierarchy">
+<ANCHOR id="GstRTSPServer.description" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.description">
+<ANCHOR id="GstRTSPServer.functions_details" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.functions_details">
+<ANCHOR id="gst-rtsp-server-new" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-new">
+<ANCHOR id="gst-rtsp-server-get-address" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-get-address">
+<ANCHOR id="gst-rtsp-server-set-address" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-set-address">
+<ANCHOR id="gst-rtsp-server-get-service" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-get-service">
+<ANCHOR id="gst-rtsp-server-set-service" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-set-service">
+<ANCHOR id="gst-rtsp-server-get-backlog" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-get-backlog">
+<ANCHOR id="gst-rtsp-server-set-backlog" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-set-backlog">
+<ANCHOR id="gst-rtsp-server-get-bound-port" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-get-bound-port">
+<ANCHOR id="gst-rtsp-server-get-mount-points" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-get-mount-points">
+<ANCHOR id="gst-rtsp-server-set-mount-points" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-set-mount-points">
+<ANCHOR id="gst-rtsp-server-get-session-pool" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-get-session-pool">
+<ANCHOR id="gst-rtsp-server-set-session-pool" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-set-session-pool">
+<ANCHOR id="gst-rtsp-server-get-thread-pool" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-get-thread-pool">
+<ANCHOR id="gst-rtsp-server-set-thread-pool" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-set-thread-pool">
+<ANCHOR id="gst-rtsp-server-get-auth" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-get-auth">
+<ANCHOR id="gst-rtsp-server-set-auth" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-set-auth">
+<ANCHOR id="gst-rtsp-server-transfer-connection" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-transfer-connection">
+<ANCHOR id="gst-rtsp-server-io-func" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-io-func">
+<ANCHOR id="gst-rtsp-server-create-socket" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-create-socket">
+<ANCHOR id="gst-rtsp-server-create-source" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-create-source">
+<ANCHOR id="gst-rtsp-server-attach" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-attach">
+<ANCHOR id="GstRTSPServerClientFilterFunc" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServerClientFilterFunc">
+<ANCHOR id="gst-rtsp-server-client-filter" href="gst-rtsp-server-1.0/GstRTSPServer.html#gst-rtsp-server-client-filter">
+<ANCHOR id="GstRTSPServer.other_details" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.other_details">
+<ANCHOR id="GstRTSPServer-struct" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer-struct">
+<ANCHOR id="GstRTSPServerClass" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServerClass">
+<ANCHOR id="GstRTSPServer.property-details" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.property-details">
+<ANCHOR id="GstRTSPServer--address" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer--address">
+<ANCHOR id="GstRTSPServer--backlog" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer--backlog">
+<ANCHOR id="GstRTSPServer--bound-port" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer--bound-port">
+<ANCHOR id="GstRTSPServer--mount-points" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer--mount-points">
+<ANCHOR id="GstRTSPServer--service" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer--service">
+<ANCHOR id="GstRTSPServer--session-pool" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer--session-pool">
+<ANCHOR id="GstRTSPServer.signal-details" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.signal-details">
+<ANCHOR id="GstRTSPServer-client-connected" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer-client-connected">
+<ANCHOR id="GstRTSPServer.see-also" href="gst-rtsp-server-1.0/GstRTSPServer.html#GstRTSPServer.see-also">
+<ANCHOR id="GstRTSPClient" href="gst-rtsp-server-1.0/GstRTSPClient.html">
+<ANCHOR id="GstRTSPClient.functions" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.functions">
+<ANCHOR id="GstRTSPClient.properties" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.properties">
+<ANCHOR id="GstRTSPClient.signals" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.signals">
+<ANCHOR id="GstRTSPClient.other" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.other">
+<ANCHOR id="GstRTSPClient.object-hierarchy" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.object-hierarchy">
+<ANCHOR id="GstRTSPClient.description" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.description">
+<ANCHOR id="GstRTSPClient.functions_details" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.functions_details">
+<ANCHOR id="gst-rtsp-client-new" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-new">
+<ANCHOR id="gst-rtsp-client-close" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-close">
+<ANCHOR id="gst-rtsp-client-get-session-pool" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-get-session-pool">
+<ANCHOR id="gst-rtsp-client-set-session-pool" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-set-session-pool">
+<ANCHOR id="gst-rtsp-client-get-mount-points" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-get-mount-points">
+<ANCHOR id="gst-rtsp-client-set-mount-points" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-set-mount-points">
+<ANCHOR id="gst-rtsp-client-get-auth" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-get-auth">
+<ANCHOR id="gst-rtsp-client-set-auth" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-set-auth">
+<ANCHOR id="gst-rtsp-client-get-thread-pool" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-get-thread-pool">
+<ANCHOR id="gst-rtsp-client-set-thread-pool" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-set-thread-pool">
+<ANCHOR id="gst-rtsp-client-get-connection" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-get-connection">
+<ANCHOR id="gst-rtsp-client-set-connection" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-set-connection">
+<ANCHOR id="gst-rtsp-client-attach" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-attach">
+<ANCHOR id="GstRTSPClientSendFunc" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClientSendFunc">
+<ANCHOR id="gst-rtsp-client-set-send-func" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-set-send-func">
+<ANCHOR id="gst-rtsp-client-handle-message" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-handle-message">
+<ANCHOR id="gst-rtsp-client-send-message" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-send-message">
+<ANCHOR id="GstRTSPClientSessionFilterFunc" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClientSessionFilterFunc">
+<ANCHOR id="gst-rtsp-client-session-filter" href="gst-rtsp-server-1.0/GstRTSPClient.html#gst-rtsp-client-session-filter">
+<ANCHOR id="GstRTSPClient.other_details" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.other_details">
+<ANCHOR id="GstRTSPClient-struct" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-struct">
+<ANCHOR id="GstRTSPClientClass" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClientClass">
+<ANCHOR id="GstRTSPClient.property-details" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.property-details">
+<ANCHOR id="GstRTSPClient--drop-backlog" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient--drop-backlog">
+<ANCHOR id="GstRTSPClient--mount-points" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient--mount-points">
+<ANCHOR id="GstRTSPClient--session-pool" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient--session-pool">
+<ANCHOR id="GstRTSPClient.signal-details" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.signal-details">
+<ANCHOR id="GstRTSPClient-closed" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-closed">
+<ANCHOR id="GstRTSPClient-describe-request" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-describe-request">
+<ANCHOR id="GstRTSPClient-get-parameter-request" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-get-parameter-request">
+<ANCHOR id="GstRTSPClient-handle-response" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-handle-response">
+<ANCHOR id="GstRTSPClient-new-session" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-new-session">
+<ANCHOR id="GstRTSPClient-options-request" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-options-request">
+<ANCHOR id="GstRTSPClient-pause-request" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-pause-request">
+<ANCHOR id="GstRTSPClient-play-request" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-play-request">
+<ANCHOR id="GstRTSPClient-send-message" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-send-message">
+<ANCHOR id="GstRTSPClient-set-parameter-request" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-set-parameter-request">
+<ANCHOR id="GstRTSPClient-setup-request" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-setup-request">
+<ANCHOR id="GstRTSPClient-teardown-request" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient-teardown-request">
+<ANCHOR id="GstRTSPClient.see-also" href="gst-rtsp-server-1.0/GstRTSPClient.html#GstRTSPClient.see-also">
+<ANCHOR id="gst-rtsp-server-GstRTSPContext" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html">
+<ANCHOR id="gst-rtsp-server-GstRTSPContext.functions" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html#gst-rtsp-server-GstRTSPContext.functions">
+<ANCHOR id="gst-rtsp-server-GstRTSPContext.other" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html#gst-rtsp-server-GstRTSPContext.other">
+<ANCHOR id="gst-rtsp-server-GstRTSPContext.description" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html#gst-rtsp-server-GstRTSPContext.description">
+<ANCHOR id="gst-rtsp-server-GstRTSPContext.functions_details" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html#gst-rtsp-server-GstRTSPContext.functions_details">
+<ANCHOR id="gst-rtsp-context-get-current" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-get-current">
+<ANCHOR id="gst-rtsp-context-push-current" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-push-current">
+<ANCHOR id="gst-rtsp-context-pop-current" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-pop-current">
+<ANCHOR id="gst-rtsp-server-GstRTSPContext.other_details" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html#gst-rtsp-server-GstRTSPContext.other_details">
+<ANCHOR id="GstRTSPContext" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html#GstRTSPContext">
+<ANCHOR id="gst-rtsp-server-GstRTSPContext.see-also" href="gst-rtsp-server-1.0/gst-rtsp-server-GstRTSPContext.html#gst-rtsp-server-GstRTSPContext.see-also">
+<ANCHOR id="GstRTSPMountPoints" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html">
+<ANCHOR id="GstRTSPMountPoints.functions" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#GstRTSPMountPoints.functions">
+<ANCHOR id="GstRTSPMountPoints.other" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#GstRTSPMountPoints.other">
+<ANCHOR id="GstRTSPMountPoints.object-hierarchy" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#GstRTSPMountPoints.object-hierarchy">
+<ANCHOR id="GstRTSPMountPoints.description" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#GstRTSPMountPoints.description">
+<ANCHOR id="GstRTSPMountPoints.functions_details" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#GstRTSPMountPoints.functions_details">
+<ANCHOR id="gst-rtsp-mount-points-new" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#gst-rtsp-mount-points-new">
+<ANCHOR id="gst-rtsp-mount-points-add-factory" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#gst-rtsp-mount-points-add-factory">
+<ANCHOR id="gst-rtsp-mount-points-remove-factory" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#gst-rtsp-mount-points-remove-factory">
+<ANCHOR id="gst-rtsp-mount-points-match" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#gst-rtsp-mount-points-match">
+<ANCHOR id="gst-rtsp-mount-points-make-path" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#gst-rtsp-mount-points-make-path">
+<ANCHOR id="GstRTSPMountPoints.other_details" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#GstRTSPMountPoints.other_details">
+<ANCHOR id="GstRTSPMountPoints-struct" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#GstRTSPMountPoints-struct">
+<ANCHOR id="GstRTSPMountPointsClass" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#GstRTSPMountPointsClass">
+<ANCHOR id="GstRTSPMountPoints.see-also" href="gst-rtsp-server-1.0/GstRTSPMountPoints.html#GstRTSPMountPoints.see-also">
+<ANCHOR id="GstRTSPMediaFactory" href="gst-rtsp-server-1.0/GstRTSPMediaFactory.html">
+<ANCHOR id="GstRTSPMediaFactory.functions" href="gst-rtsp-server-1.0/GstRTSPMediaFactory.html#GstRTSPMediaFactory.functions">
+<ANCHOR id="GstRTSPMediaFactory.properties" href="gst-rtsp-server-1.0/GstRTSPMediaFactory.html#GstRTSPMediaFactory.properties">
+<ANCHOR id="GstRTSPMediaFactory.signals" href="gst-rtsp-server-1.0/GstRTSPMediaFactory.html#GstRTSPMediaFactory.signals">
+<ANCHOR id="GstRTSPMediaFactory.other" href="gst-rtsp-server-1.0/GstRTSPMediaFactory.html#GstRTSPMediaFactory.other">
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+<ANCHOR id="GstRTSPSessionFilterFunc" href="gst-rtsp-server-1.0/GstRTSPSession.html#GstRTSPSessionFilterFunc">
+<ANCHOR id="gst-rtsp-session-filter" href="gst-rtsp-server-1.0/GstRTSPSession.html#gst-rtsp-session-filter">
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+<title>GStreamer RTSP Server Reference Manual: Object Hierarchy</title>
+<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
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+</tr></table>
+<div class="chapter">
+<div class="titlepage"><div><div><h1 class="title">
+<a name="rtsp-server-hierarchy"></a>Object Hierarchy</h1></div></div></div>
+<pre class="screen">
+ <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+ <span class="lineart">├──</span> <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
+ <span class="lineart">├──</span> <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
+ <span class="lineart">├──</span> <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
+ <span class="lineart">├──</span> <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
+ <span class="lineart">├──</span> <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
+ <span class="lineart">├──</span> <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
+ <span class="lineart">├──</span> <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
+ <span class="lineart">├──</span> <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
+ <span class="lineart">╰──</span> <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
+</pre>
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+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+typedef struct
+{
+ gboolean white;
+ GstClockTime timestamp;
+} MyContext;
+
+/* called when we need to give data to appsrc */
+static void
+need_data (GstElement * appsrc, guint unused, MyContext * ctx)
+{
+ GstBuffer *buffer;
+ guint size;
+ GstFlowReturn ret;
+
+ size = 385 * 288 * 2;
+
+ buffer = gst_buffer_new_allocate (NULL, size, NULL);
+
+ /* this makes the image black/white */
+ gst_buffer_memset (buffer, 0, ctx->white ? 0xff : 0x0, size);
+
+ ctx->white = !ctx->white;
+
+ /* increment the timestamp every 1/2 second */
+ GST_BUFFER_PTS (buffer) = ctx->timestamp;
+ GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND, 2);
+ ctx->timestamp += GST_BUFFER_DURATION (buffer);
+
+ g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
+}
+
+/* called when a new media pipeline is constructed. We can query the
+ * pipeline and configure our appsrc */
+static void
+media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media,
+ gpointer user_data)
+{
+ GstElement *element, *appsrc;
+ MyContext *ctx;
+
+ /* get the element used for providing the streams of the media */
+ element = gst_rtsp_media_get_element (media);
+
+ /* get our appsrc, we named it 'mysrc' with the name property */
+ appsrc = gst_bin_get_by_name_recurse_up (GST_BIN (element), "mysrc");
+
+ /* this instructs appsrc that we will be dealing with timed buffer */
+ gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
+ /* configure the caps of the video */
+ g_object_set (G_OBJECT (appsrc), "caps",
+ gst_caps_new_simple ("video/x-raw",
+ "format", G_TYPE_STRING, "RGB16",
+ "width", G_TYPE_INT, 384,
+ "height", G_TYPE_INT, 288,
+ "framerate", GST_TYPE_FRACTION, 0, 1, NULL), NULL);
+
+ ctx = g_new0 (MyContext, 1);
+ ctx->white = FALSE;
+ ctx->timestamp = 0;
+ /* make sure ther datais freed when the media is gone */
+ g_object_set_data_full (G_OBJECT (media), "my-extra-data", ctx,
+ (GDestroyNotify) g_free);
+
+ /* install the callback that will be called when a buffer is needed */
+ g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "( appsrc name=mysrc ! videoconvert ! x264enc ! rtph264pay name=pay0 pt=96 )");
+
+ /* notify when our media is ready, This is called whenever someone asks for
+ * the media and a new pipeline with our appsrc is created */
+ g_signal_connect (factory, "media-configure", (GCallback) media_configure,
+ NULL);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mounts anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+static gboolean
+remove_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
+ GstRTSPServer * server)
+{
+ return GST_RTSP_FILTER_REMOVE;
+}
+
+static gboolean
+remove_sessions (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ g_print ("removing all sessions\n");
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_filter (pool,
+ (GstRTSPSessionPoolFilterFunc) remove_func, server);
+ g_object_unref (pool);
+
+ return FALSE;
+}
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ gchar *basic;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mounts for this server, every server has a default mapper object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* allow user and admin to access this resource */
+ gst_rtsp_media_factory_add_role (factory, "user",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_media_factory_add_role (factory, "admin",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+ /* admin2 can look at the media but not construct so he gets a
+ * 401 Unauthorized */
+ gst_rtsp_media_factory_add_role (factory, "admin2",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, FALSE, NULL);
+
+ /* make another factory */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=30/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 )");
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test2", factory);
+
+ /* allow admin2 to access this resource */
+ /* user and admin have no permissions so they can't even see the
+ * media and get a 404 Not Found */
+ gst_rtsp_media_factory_add_role (factory, "admin2",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* make a new authentication manager */
+ auth = gst_rtsp_auth_new ();
+
+ /* make default token, it has the same permissions as admin2 */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "admin2", NULL);
+ gst_rtsp_auth_set_default_token (auth, token);
+ gst_rtsp_token_unref (token);
+
+ /* make user token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ basic = gst_rtsp_auth_make_basic ("user", "password");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* make admin token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "admin", NULL);
+ basic = gst_rtsp_auth_make_basic ("admin", "power");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* make admin2 token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "admin2", NULL);
+ basic = gst_rtsp_auth_make_basic ("admin2", "power2");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* set as the server authentication manager */
+ gst_rtsp_server_set_auth (server, auth);
+ g_object_unref (auth);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+ g_timeout_add_seconds (10, (GSourceFunc) remove_sessions, server);
+
+ /* start serving */
+ g_print ("stream with user:password ready at rtsp://127.0.0.1:8554/test\n");
+ g_print ("stream with admin:power ready at rtsp://127.0.0.1:8554/test\n");
+ g_print ("stream with admin2:power2 ready at rtsp://127.0.0.1:8554/test2\n");
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/* Runs a pipeline and clasifies the media pipelines based on the
+ * authenticated user.
+ *
+ * This test requires 2 cpu cgroups to exist named 'user' and 'admin'.
+ * The rtsp server should have permission to add its threads to the
+ * cgroups.
+ *
+ * sudo cgcreate -t uid:gid -g cpu:/user
+ * sudo cgcreate -t uid:gid -g cpu:/admin
+ *
+ * With -t you can give the user and group access to the task file to
+ * write the thread ids. The user running the server can be used.
+ *
+ * Then you would want to change the cpu shares assigned to each group:
+ *
+ * sudo cgset -r cpu.shares=100 user
+ * sudo cgset -r cpu.shares=1024 admin
+ *
+ * Then start clients for 'user' until the stream is degraded because of
+ * lack of CPU. Then start a client for 'admin' and check that the stream
+ * is not degraded.
+ */
+
+#include <libcgroup.h>
+
+#include <gst/gst.h>
+#include <gst/rtsp-server/rtsp-server.h>
+
+typedef struct _GstRTSPCGroupPool GstRTSPCGroupPool;
+typedef struct _GstRTSPCGroupPoolClass GstRTSPCGroupPoolClass;
+
+#define GST_TYPE_RTSP_CGROUP_POOL (gst_rtsp_cgroup_pool_get_type ())
+#define GST_IS_RTSP_CGROUP_POOL(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CGROUP_POOL))
+#define GST_IS_RTSP_CGROUP_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CGROUP_POOL))
+#define GST_RTSP_CGROUP_POOL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CGROUP_POOL, GstRTSPCGroupPoolClass))
+#define GST_RTSP_CGROUP_POOL(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CGROUP_POOL, GstRTSPCGroupPool))
+#define GST_RTSP_CGROUP_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CGROUP_POOL, GstRTSPCGroupPoolClass))
+#define GST_RTSP_CGROUP_POOL_CAST(obj) ((GstRTSPCGroupPool*)(obj))
+#define GST_RTSP_CGROUP_POOL_CLASS_CAST(klass) ((GstRTSPCGroupPoolClass*)(klass))
+
+struct _GstRTSPCGroupPool
+{
+ GstRTSPThreadPool parent;
+
+ struct cgroup *user;
+ struct cgroup *admin;
+};
+
+struct _GstRTSPCGroupPoolClass
+{
+ GstRTSPThreadPoolClass parent_class;
+};
+
+static GQuark thread_cgroup;
+
+static void gst_rtsp_cgroup_pool_finalize (GObject * obj);
+
+static void default_thread_enter (GstRTSPThreadPool * pool,
+ GstRTSPThread * thread);
+static void default_configure_thread (GstRTSPThreadPool * pool,
+ GstRTSPThread * thread, GstRTSPContext * ctx);
+
+static GType gst_rtsp_cgroup_pool_get_type (void);
+
+G_DEFINE_TYPE (GstRTSPCGroupPool, gst_rtsp_cgroup_pool,
+ GST_TYPE_RTSP_THREAD_POOL);
+
+static void
+gst_rtsp_cgroup_pool_class_init (GstRTSPCGroupPoolClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstRTSPThreadPoolClass *tpool_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ tpool_class = GST_RTSP_THREAD_POOL_CLASS (klass);
+
+ gobject_class->finalize = gst_rtsp_cgroup_pool_finalize;
+
+ tpool_class->configure_thread = default_configure_thread;
+ tpool_class->thread_enter = default_thread_enter;
+
+ thread_cgroup = g_quark_from_string ("cgroup.pool.thread.cgroup");
+
+ cgroup_init ();
+}
+
+static void
+gst_rtsp_cgroup_pool_init (GstRTSPCGroupPool * pool)
+{
+ pool->user = cgroup_new_cgroup ("user");
+ if (cgroup_add_controller (pool->user, "cpu") == NULL)
+ g_error ("Failed to add cpu controller to user cgroup");
+ pool->admin = cgroup_new_cgroup ("admin");
+ if (cgroup_add_controller (pool->admin, "cpu") == NULL)
+ g_error ("Failed to add cpu controller to admin cgroup");
+}
+
+static void
+gst_rtsp_cgroup_pool_finalize (GObject * obj)
+{
+ GstRTSPCGroupPool *pool = GST_RTSP_CGROUP_POOL (obj);
+
+ GST_INFO ("finalize pool %p", pool);
+
+ cgroup_free (&pool->user);
+ cgroup_free (&pool->admin);
+
+ G_OBJECT_CLASS (gst_rtsp_cgroup_pool_parent_class)->finalize (obj);
+}
+
+static void
+default_thread_enter (GstRTSPThreadPool * pool, GstRTSPThread * thread)
+{
+ struct cgroup *cgroup;
+
+ cgroup = gst_mini_object_get_qdata (GST_MINI_OBJECT (thread), thread_cgroup);
+ if (cgroup) {
+ gint res = 0;
+
+ res = cgroup_attach_task (cgroup);
+
+ if (res != 0)
+ GST_ERROR ("error: %d (%s)", res, cgroup_strerror (res));
+ }
+}
+
+static void
+default_configure_thread (GstRTSPThreadPool * pool,
+ GstRTSPThread * thread, GstRTSPContext * ctx)
+{
+ GstRTSPCGroupPool *cpool = GST_RTSP_CGROUP_POOL (pool);
+ const gchar *cls;
+ struct cgroup *cgroup;
+
+ if (ctx->token)
+ cls = gst_rtsp_token_get_string (ctx->token, "cgroup.pool.media.class");
+ else
+ cls = NULL;
+
+ GST_DEBUG ("manage cgroup %s", cls);
+
+ if (!g_strcmp0 (cls, "admin"))
+ cgroup = cpool->admin;
+ else
+ cgroup = cpool->user;
+
+ /* attach the cgroup to the thread */
+ gst_mini_object_set_qdata (GST_MINI_OBJECT (thread), thread_cgroup,
+ cgroup, NULL);
+}
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ gchar *basic;
+ GstRTSPThreadPool *thread_pool;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mounts for this server, every server has a default mapper object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=640,height=480,framerate=50/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* allow user and admin to access this resource */
+ gst_rtsp_media_factory_add_role (factory, "user",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_media_factory_add_role (factory, "admin",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* make a new authentication manager */
+ auth = gst_rtsp_auth_new ();
+
+ /* make user token */
+ token = gst_rtsp_token_new ("cgroup.pool.media.class", G_TYPE_STRING, "user",
+ "media.factory.role", G_TYPE_STRING, "user", NULL);
+ basic = gst_rtsp_auth_make_basic ("user", "password");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* make admin token */
+ token = gst_rtsp_token_new ("cgroup.pool.media.class", G_TYPE_STRING, "admin",
+ "media.factory.role", G_TYPE_STRING, "admin", NULL);
+ basic = gst_rtsp_auth_make_basic ("admin", "power");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* set as the server authentication manager */
+ gst_rtsp_server_set_auth (server, auth);
+ g_object_unref (auth);
+
+ thread_pool = g_object_new (GST_TYPE_RTSP_CGROUP_POOL, NULL);
+ gst_rtsp_server_set_thread_pool (server, thread_pool);
+ g_object_unref (thread_pool);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving */
+ g_print ("stream with user:password ready at rtsp://127.0.0.1:8554/test\n");
+ g_print ("stream with admin:power ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ if (argc < 2) {
+ g_print ("usage: %s <launch line> \n"
+ "example: %s \"( videotestsrc ! x264enc ! rtph264pay name=pay0 pt=96 )\"\n",
+ argv[0], argv[0]);
+ return -1;
+ }
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, argv[1]);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ gchar *str;
+
+ gst_init (&argc, &argv);
+
+ if (argc < 2) {
+ g_message ("usage: %s <filename.mp4>", argv[0]);
+ return -1;
+ }
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ str = g_strdup_printf ("( "
+ "filesrc location=%s ! qtdemux name=d "
+ "d. ! queue ! rtph264pay pt=96 name=pay0 "
+ "d. ! queue ! rtpmp4apay pt=97 name=pay1 " ")", argv[1]);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, str);
+ g_free (str);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+
+static gboolean
+timeout (GstRTSPServer * server, gboolean ignored)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAddressPool *pool;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+
+ /* make a new address pool */
+ pool = gst_rtsp_address_pool_new ();
+ gst_rtsp_address_pool_add_range (pool,
+ "224.3.0.0", "224.3.0.10", 5000, 5010, 16);
+ gst_rtsp_media_factory_set_address_pool (factory, pool);
+ /* only allow multicast */
+ gst_rtsp_media_factory_set_protocols (factory,
+ GST_RTSP_LOWER_TRANS_UDP_MCAST);
+ g_object_unref (pool);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+
+static gboolean
+timeout (GstRTSPServer * server, gboolean ignored)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+static void
+media_constructed (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+ guint i, n_streams;
+
+ n_streams = gst_rtsp_media_n_streams (media);
+
+ for (i = 0; i < n_streams; i++) {
+ GstRTSPAddressPool *pool;
+ GstRTSPStream *stream;
+ gchar *min, *max;
+
+ stream = gst_rtsp_media_get_stream (media, i);
+
+ /* make a new address pool */
+ pool = gst_rtsp_address_pool_new ();
+
+ min = g_strdup_printf ("224.3.0.%d", (2 * i) + 1);
+ max = g_strdup_printf ("224.3.0.%d", (2 * i) + 2);
+ gst_rtsp_address_pool_add_range (pool, min, max,
+ 5000 + (10 * i), 5010 + (10 * i), 1);
+ g_free (min);
+ g_free (max);
+
+ gst_rtsp_stream_set_address_pool (stream, pool);
+ g_object_unref (pool);
+ }
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+
+ g_signal_connect (factory, "media-constructed", (GCallback)
+ media_constructed, NULL);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ gchar *str;
+
+ gst_init (&argc, &argv);
+
+ if (argc < 2) {
+ g_message ("usage: %s <filename.ogg>", argv[0]);
+ return -1;
+ }
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ str = g_strdup_printf ("( "
+ "filesrc location=%s ! oggdemux name=d "
+ "d. ! queue ! rtptheorapay name=pay0 pt=96 "
+ "d. ! queue ! rtpvorbispay name=pay1 pt=97 " ")", argv[1]);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, str);
+ g_free (str);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc is-live=1 ! x264enc ! rtph264pay name=pay0 pt=96 )");
+
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2009 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+
+static gboolean
+timeout (GstRTSPServer * server, gboolean ignored)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+static void
+media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+ gst_rtsp_media_set_reusable (media, TRUE);
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ gchar *str;
+
+ gst_init (&argc, &argv);
+
+ if (argc < 2) {
+ g_message ("usage: %s <filename.sdp>", argv[0]);
+ return -1;
+ }
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+
+ str =
+ g_strdup_printf ("( filesrc location=%s ! sdpdemux name=dynpay0 )",
+ argv[1]);
+ gst_rtsp_media_factory_set_launch (factory, str);
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ g_signal_connect (factory, "media-configure", (GCallback) media_configure,
+ NULL);
+ g_free (str);
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ gst_rtsp_server_attach (server, NULL);
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving */
+ g_main_loop_run (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+#include <gst/rtsp-server/rtsp-media-factory-uri.h>
+
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+static gboolean
+remove_map (GstRTSPServer * server)
+{
+ GstRTSPMountPoints *mounts;
+
+ g_print ("removing /test mount point\n");
+ mounts = gst_rtsp_server_get_mount_points (server);
+ gst_rtsp_mount_points_remove_factory (mounts, "/test");
+ g_object_unref (mounts);
+
+ return FALSE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactoryURI *factory;
+
+ gst_init (&argc, &argv);
+
+ if (argc < 2) {
+ g_message ("usage: %s <uri>", argv[0]);
+ return -1;
+ }
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a URI media factory for a test stream. */
+ factory = gst_rtsp_media_factory_uri_new ();
+ /* when using GStreamer as a client, one can use the gst payloader, which is
+ * more efficient when there is no payloader for the compressed format */
+ /* g_object_set (factory, "use-gstpay", TRUE, NULL); */
+ gst_rtsp_media_factory_uri_set_uri (factory, argv[1]);
+ /* if you want multiple clients to see the same video, set the shared property
+ * to TRUE */
+ /* gst_rtsp_media_factory_set_shared ( GST_RTSP_MEDIA_FACTORY (factory), TRUE); */
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test",
+ GST_RTSP_MEDIA_FACTORY (factory));
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ /* do session cleanup every 2 seconds */
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+ /* remove the mount point after 10 seconds, new clients won't be able to use the
+ * /test url anymore */
+ g_timeout_add_seconds (10, (GSourceFunc) remove_map, server);
+
+ /* start serving */
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+/* define this if you want the resource to only be available when using
+ * user/password as the password */
+#undef WITH_AUTH
+
+/* define this if you want the server to use TLS (it will also need WITH_AUTH
+ * to be defined) */
+#undef WITH_TLS
+
+/* this timeout is periodically run to clean up the expired sessions from the
+ * pool. This needs to be run explicitly currently but might be done
+ * automatically as part of the mainloop. */
+static gboolean
+timeout (GstRTSPServer * server, gboolean ignored)
+{
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ gst_rtsp_session_pool_cleanup (pool);
+ g_object_unref (pool);
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+#ifdef WITH_AUTH
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ gchar *basic;
+ GstRTSPPermissions *permissions;
+#endif
+#ifdef WITH_TLS
+ GTlsCertificate *cert;
+ GError *error = NULL;
+#endif
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+#ifdef WITH_AUTH
+ /* make a new authentication manager. it can be added to control access to all
+ * the factories on the server or on individual factories. */
+ auth = gst_rtsp_auth_new ();
+#ifdef WITH_TLS
+ cert = g_tls_certificate_new_from_pem ("-----BEGIN CERTIFICATE-----"
+ "MIICJjCCAY+gAwIBAgIBBzANBgkqhkiG9w0BAQUFADCBhjETMBEGCgmSJomT8ixk"
+ "ARkWA0NPTTEXMBUGCgmSJomT8ixkARkWB0VYQU1QTEUxHjAcBgNVBAsTFUNlcnRp"
+ "ZmljYXRlIEF1dGhvcml0eTEXMBUGA1UEAxMOY2EuZXhhbXBsZS5jb20xHTAbBgkq"
+ "hkiG9w0BCQEWDmNhQGV4YW1wbGUuY29tMB4XDTExMDExNzE5NDcxN1oXDTIxMDEx"
+ "NDE5NDcxN1owSzETMBEGCgmSJomT8ixkARkWA0NPTTEXMBUGCgmSJomT8ixkARkW"
+ "B0VYQU1QTEUxGzAZBgNVBAMTEnNlcnZlci5leGFtcGxlLmNvbTBcMA0GCSqGSIb3"
+ "DQEBAQUAA0sAMEgCQQDYScTxk55XBmbDM9zzwO+grVySE4rudWuzH2PpObIonqbf"
+ "hRoAalKVluG9jvbHI81eXxCdSObv1KBP1sbN5RzpAgMBAAGjIjAgMAkGA1UdEwQC"
+ "MAAwEwYDVR0lBAwwCgYIKwYBBQUHAwEwDQYJKoZIhvcNAQEFBQADgYEAYx6fMqT1"
+ "Gvo0jq88E8mc+bmp4LfXD4wJ7KxYeadQxt75HFRpj4FhFO3DOpVRFgzHlOEo3Fwk"
+ "PZOKjvkT0cbcoEq5whLH25dHoQxGoVQgFyAP5s+7Vp5AlHh8Y/vAoXeEVyy/RCIH"
+ "QkhUlAflfDMcrrYjsmwoOPSjhx6Mm/AopX4="
+ "-----END CERTIFICATE-----"
+ "-----BEGIN PRIVATE KEY-----"
+ "MIIBVAIBADANBgkqhkiG9w0BAQEFAASCAT4wggE6AgEAAkEA2EnE8ZOeVwZmwzPc"
+ "88DvoK1ckhOK7nVrsx9j6TmyKJ6m34UaAGpSlZbhvY72xyPNXl8QnUjm79SgT9bG"
+ "zeUc6QIDAQABAkBRFJZ32VbqWMP9OVwDJLiwC01AlYLnka0mIQZbT/2xq9dUc9GW"
+ "U3kiVw4lL8v/+sPjtTPCYYdzHHOyDen6znVhAiEA9qJT7BtQvRxCvGrAhr9MS022"
+ "tTdPbW829BoUtIeH64cCIQDggG5i48v7HPacPBIH1RaSVhXl8qHCpQD3qrIw3FMw"
+ "DwIga8PqH5Sf5sHedy2+CiK0V4MRfoU4c3zQ6kArI+bEgSkCIQCLA1vXBiE31B5s"
+ "bdHoYa1BXebfZVd+1Hd95IfEM5mbRwIgSkDuQwV55BBlvWph3U8wVIMIb4GStaH8"
+ "W535W8UBbEg=" "-----END PRIVATE KEY-----", -1, &error);
+ if (cert == NULL) {
+ g_printerr ("failed to parse PEM: %s\n", error->message);
+ return -1;
+ }
+ gst_rtsp_auth_set_tls_certificate (auth, cert);
+ g_object_unref (cert);
+#endif
+
+ /* make user token */
+ token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ basic = gst_rtsp_auth_make_basic ("user", "password");
+ gst_rtsp_auth_add_basic (auth, basic, token);
+ g_free (basic);
+ gst_rtsp_token_unref (token);
+
+ /* configure in the server */
+ gst_rtsp_server_set_auth (server, auth);
+#endif
+
+ /* get the mount points for this server, every server has a default object
+ * that be used to map uri mount points to media factories */
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ /* make a media factory for a test stream. The default media factory can use
+ * gst-launch syntax to create pipelines.
+ * any launch line works as long as it contains elements named pay%d. Each
+ * element with pay%d names will be a stream */
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory, "( "
+ "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+ "x264enc ! rtph264pay name=pay0 pt=96 "
+ "audiotestsrc ! audio/x-raw,rate=8000 ! "
+ "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+#ifdef WITH_AUTH
+ /* add permissions for the user media role */
+ permissions = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (permissions, "user",
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_media_factory_set_permissions (factory, permissions);
+ gst_rtsp_permissions_unref (permissions);
+#ifdef WITH_TLS
+ gst_rtsp_media_factory_set_profiles (factory, GST_RTSP_PROFILE_SAVP);
+#endif
+#endif
+
+ /* attach the test factory to the /test url */
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ /* don't need the ref to the mapper anymore */
+ g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if (gst_rtsp_server_attach (server, NULL) == 0)
+ goto failed;
+
+ /* add a timeout for the session cleanup */
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+ /* start serving, this never stops */
+#ifdef WITH_TLS
+ g_print ("stream ready at rtsps://127.0.0.1:8554/test\n");
+#else
+ g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+#endif
+ g_main_loop_run (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <glib-unix.h>
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server-wfd.h>
+#include <gst/rtsp-server/rtsp-media-factory-wfd.h>
+
+#if 1
+#define VIDEO_PIPELINE "ximagesrc ! videorate ! videoscale ! videoconvert ! " \
+ "video/x-raw,width=640,height=480,framerate=30/1 ! " \
+ "x264enc aud=false byte-stream=true bitrate=512 ! video/x-h264,profile=baseline ! mpegtsmux wfd-mode=TRUE ! " \
+ "rtpmp2tpay name=pay0 pt=33"
+#define AUDIO_PIPELINE "pulsesrc device=alsa_output.pci-0000_00_1b.0.analog-stereo.monitor ! audioconvert ! " \
+ "faac ! mpegtsmux wfd-mode=TRUE ! " \
+ "rtpmp2tpay name=pay0 pt=33"
+#else
+#define VIDEO_PIPELINE "ximagesrc ! videoscale ! videoconvert ! " \
+ "video/x-raw,width=640,height=480,framerate=60/1 ! " \
+ "x264enc aud=false byte-stream=true bitrate=512 ! video/x-h264,profile=baseline ! mpegtsmux name=mux " \
+ "pulsesrc device=alsa_output.pci-0000_00_1b.0.analog-stereo.monitor ! audioconvert ! " \
+ "faac ! mux. mux. ! " \
+ "rtpmp2tpay name=pay0 pt=33"
+#if 0
+#define VIDEO_PIPELINE "ximagesrc startx=0 starty=0 endx=1919 endy=1079 ! videorate ! videoscale ! video/x-raw,width=1280,height=720,framerate=30/1 ! videoconvert ! " \
+ "x264enc aud=false byte-stream=true bitrate=512 ! video/x-h264,profile=baseline ! mpegtsmux name=mux wfd-mode=TRUE " \
+ "pulsesrc device=alsa_output.pci-0000_00_1b.0.analog-stereo.monitor ! audioconvert ! " \
+ "faac ! mux. mux. ! " \
+ "rtpmp2tpay name=pay0 pt=33"
+#endif
+/*
+#define VIDEO_PIPELINE "ximagesrc do-timestamp=true ! videoscale ! video/x-raw,width=1280,height=720,framerate=30/1 ! videoconvert ! " \
+ "x264enc aud=false byte-stream=true bitrate=512 ! video/x-h264,profile=baseline ! mpegtsmux name=mux wfd-mode=TRUE " \
+ "pulsesrc device=alsa_output.pci-0000_00_1b.0.analog-stereo.monitor provide-clock=false ! audioconvert ! " \
+ "faac ! mux. mux. ! " \
+ "rtpmp2tpay name=pay0 pt=33"
+*/
+#endif
+
+#define WFD_RTSP_PORT "2022"
+#define TEST_MOUNT_POINT "/wfd1.0/streamid=0"
+
+GMainLoop *loop;
+
+static gboolean
+teardown(gpointer data)
+{
+ GstRTSPWFDServer *server = NULL;
+ server = (GstRTSPWFDServer *) data;
+
+ g_print("teardown\n");
+
+ if (server == NULL) return FALSE;
+
+ gst_rtsp_wfd_server_trigger_request (GST_RTSP_SERVER(server), WFD_TRIGGER_TEARDOWN);
+
+ return FALSE;
+}
+
+int main (int argc, char *argv[])
+{
+ GstRTSPWFDServer *server;
+ guint id;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactoryWFD *factory;
+ GMainContext *context = NULL;
+ GSource * signal_handler_src = NULL;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+ context = g_main_loop_get_context(loop);
+
+ /* create a server instance */
+ server = gst_rtsp_wfd_server_new ();
+
+ gst_rtsp_server_set_address(GST_RTSP_SERVER(server), "192.168.3.100");
+ //gst_rtsp_server_set_address(GST_RTSP_SERVER(server), "192.168.49.194");
+ //gst_rtsp_server_set_address(GST_RTSP_SERVER(server), "192.168.49.20");
+ gst_rtsp_server_set_service(GST_RTSP_SERVER(server), WFD_RTSP_PORT);
+ mounts = gst_rtsp_server_get_mount_points (GST_RTSP_SERVER(server));
+
+ factory = gst_rtsp_media_factory_wfd_new ();
+
+ gst_rtsp_media_factory_set_launch (GST_RTSP_MEDIA_FACTORY(factory),
+ "( " VIDEO_PIPELINE " )");
+ g_object_ref (factory);
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, GST_RTSP_MEDIA_FACTORY(factory));
+ if (mounts) g_object_unref (mounts);
+
+ /* attach the server to the default maincontext */
+ if ((id = gst_rtsp_server_attach (GST_RTSP_SERVER_CAST(server), NULL)) == 0)
+ goto failed;
+
+ signal_handler_src = (GSource *)g_unix_signal_source_new (SIGINT);
+ /* Set callback to be called when socket is readable */
+ g_source_set_callback(signal_handler_src, teardown, server, NULL);
+ g_source_attach(signal_handler_src, context);
+
+ /* start serving */
+ g_main_loop_run (loop);
+
+ /* cleanup */
+ g_source_remove (id);
+ g_object_unref (server);
+ g_main_loop_unref (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+<Project
+ xmlns:rdf="http://www.w3.org/1999/02/22-rdf-syntax-ns#"
+ xmlns:rdfs="http://www.w3.org/2000/01/rdf-schema#"
+ xmlns="http://usefulinc.com/ns/doap#"
+ xmlns:foaf="http://xmlns.com/foaf/0.1/"
+ xmlns:admin="http://webns.net/mvcb/">
+
+ <name>GStreamer RTSP Server</name>
+ <shortname>gst-rtsp-server</shortname>
+ <homepage rdf:resource="http://gstreamer.freedesktop.org/modules/gst-rtsp-server.html" />
+ <created>1999-10-31</created>
+ <shortdesc xml:lang="en">
+RTSP server library based on GStreamer
+</shortdesc>
+ <description xml:lang="en">
+RTSP server library based on GStreamer
+ </description>
+ <category></category>
+ <bug-database rdf:resource="http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-rtsp-server" />
+ <screenshots></screenshots>
+ <mailing-list rdf:resource="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" />
+ <programming-language>C</programming-language>
+ <license rdf:resource="http://usefulinc.com/doap/licenses/lgpl" />
+ <download-page rdf:resource="http://gstreamer.freedesktop.org/download/" />
+
+ <repository>
+ <GitRepository>
+ <location rdf:resource="git://anongit.freedesktop.org/gstreamer/gst-rtsp-server"/>
+ <browse rdf:resource="http://cgit.freedesktop.org/gstreamer/gst-rtsp-server"/>
+ </GitRepository>
+</repository>
+
+ <release>
+ <Version>
+ <revision>1.4.5</revision>
+ <branch>1.4</branch>
+ <name></name>
+ <created>2014-12-18</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.4.5.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.4.4</revision>
+ <branch>1.4</branch>
+ <name></name>
+ <created>2014-11-06</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.4.4.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.4.3</revision>
+ <branch>1.4</branch>
+ <name></name>
+ <created>2014-09-24</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.4.3.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.4.2</revision>
+ <branch>1.4</branch>
+ <name></name>
+ <created>2014-09-19</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.4.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.4.1</revision>
+ <branch>1.4</branch>
+ <name></name>
+ <created>2014-08-27</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.4.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.4.0</revision>
+ <branch>1.4</branch>
+ <name></name>
+ <created>2014-07-19</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.4.0.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.3.91</revision>
+ <branch>1.3</branch>
+ <name></name>
+ <created>2014-07-11</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.91.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.3.90</revision>
+ <branch>1.3</branch>
+ <name></name>
+ <created>2014-06-28</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.3.3</revision>
+ <branch>1.3</branch>
+ <name></name>
+ <created>2014-06-22</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.3.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.3.2</revision>
+ <branch>1.3</branch>
+ <name></name>
+ <created>2014-05-21</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.3.1</revision>
+ <branch>1.3</branch>
+ <name></name>
+ <created>2014-05-03</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
+ <revision>1.1.90</revision>
+ <branch>1.1</branch>
+ <name></name>
+ <created>2014-02-09</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.1.90.tar.xz" />
+ </Version>
+ </release>
+
+ <maintainer>
+ <foaf:Person>
+ <foaf:name>Wim Taymans</foaf:name>
+ <foaf:mbox_sha1sum>0d93fde052812d51a05fd86de9bdbf674423daa2</foaf:mbox_sha1sum>
+ </foaf:Person>
+ </maintainer>
+
+</Project>
--- /dev/null
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+</manifest>
--- /dev/null
+SUBDIRS = rtsp-server
--- /dev/null
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+ rtsp-media.h \
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+ rtsp-mount-points.h \
+ rtsp-permissions.h \
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+ rtsp-token.h \
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+ rtsp-server-wfd.h \
+ rtsp-server.h \
+ gstwfdmessage.h
+
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+ rtsp-media.c \
+ rtsp-media-factory.c \
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+ rtsp-media-factory-uri.c \
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+ rtsp-permissions.c \
+ rtsp-stream.c \
+ rtsp-stream-transport.c \
+ rtsp-session.c \
+ rtsp-session-media.c \
+ rtsp-session-pool.c \
+ rtsp-token.c \
+ gstwfdmessage.c \
+ rtsp-client-wfd.c \
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+ rtsp-server-wfd.c \
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+ rtsp-auth.h \
+ rtsp-address-pool.h \
+ rtsp-context.h \
+ rtsp-params.h \
+ rtsp-sdp.h \
+ rtsp-thread-pool.h \
+ rtsp-media.h \
+ rtsp-media-factory.h \
+ rtsp-media-factory-uri.h \
+ rtsp-mount-points.h \
+ rtsp-permissions.h \
+ rtsp-stream.h \
+ rtsp-stream-transport.h \
+ rtsp-session.h \
+ rtsp-session-media.h \
+ rtsp-session-pool.h \
+ rtsp-token.h \
+ rtsp-client.h \
+ rtsp-server.h
+
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+ rtsp-auth.c \
+ rtsp-address-pool.c \
+ rtsp-context.c \
+ rtsp-params.c \
+ rtsp-sdp.c \
+ rtsp-thread-pool.c \
+ rtsp-media.c \
+ rtsp-media-factory.c \
+ rtsp-media-factory-uri.c \
+ rtsp-mount-points.c \
+ rtsp-permissions.c \
+ rtsp-stream.c \
+ rtsp-stream-transport.c \
+ rtsp-session.c \
+ rtsp-session-media.c \
+ rtsp-session-pool.c \
+ rtsp-token.c \
+ rtsp-client.c \
+ rtsp-server.c
+
+noinst_HEADERS =
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+ libgstrtspserver-@GST_API_VERSION@.la
+
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+ $(c_sources)
+
+libgstrtspserver_@GST_API_VERSION@_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
+libgstrtspserver_@GST_API_VERSION@_la_LDFLAGS = $(GST_LIB_LDFLAGS) $(GST_ALL_LDFLAGS) $(GST_LT_LDFLAGS)
+libgstrtspserver_@GST_API_VERSION@_la_LIBADD = \
+ $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) \
+ -lgstrtp-@GST_API_VERSION@ -lgstrtsp-@GST_API_VERSION@ \
+ -lgstnet-@GST_API_VERSION@ \
+ -lgstsdp-@GST_API_VERSION@ \
+ -lgstapp-@GST_API_VERSION@ \
+ $(GST_LIBS) $(GIO_LIBS) $(LIBM)
+
+libgstrtspserver_@GST_API_VERSION@includedir = $(includedir)/gstreamer-@GST_API_VERSION@/gst/rtsp-server
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+@HAVE_INTROSPECTION_TRUE@gir_sources = $(patsubst %,$(srcdir)/%, $(libgstrtspserver_@GST_API_VERSION@_la_SOURCES))
+
+# INTROSPECTION_GIRDIR/INTROSPECTION_TYPELIBDIR aren't the right place to
+# install anything - we need to install inside our prefix.
+@HAVE_INTROSPECTION_TRUE@girdir = $(datadir)/gir-1.0
+@HAVE_INTROSPECTION_TRUE@gir_DATA = $(BUILT_GIRSOURCES)
+@HAVE_INTROSPECTION_TRUE@typelibsdir = $(libdir)/girepository-1.0/
+@HAVE_INTROSPECTION_TRUE@typelibs_DATA = $(BUILT_GIRSOURCES:.gir=.typelib)
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+
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+ @for dep in $?; do \
+ case '$(am__configure_deps)' in \
+ *$$dep*) \
+ ( cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh ) \
+ && { if test -f $@; then exit 0; else break; fi; }; \
+ exit 1;; \
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+ $(MKDIR_P) "$(DESTDIR)$(libgstrtspserver_@GST_API_VERSION@includedir)" || exit 1; \
+ fi; \
+ for p in $$list; do \
+ if test -f "$$p"; then d=; else d="$(srcdir)/"; fi; \
+ echo "$$d$$p"; \
+ done | $(am__base_list) | \
+ while read files; do \
+ echo " $(INSTALL_HEADER) $$files '$(DESTDIR)$(libgstrtspserver_@GST_API_VERSION@includedir)'"; \
+ $(INSTALL_HEADER) $$files "$(DESTDIR)$(libgstrtspserver_@GST_API_VERSION@includedir)" || exit $$?; \
+ done
+
+uninstall-libgstrtspserver_@GST_API_VERSION@includeHEADERS:
+ @$(NORMAL_UNINSTALL)
+ @list='$(libgstrtspserver_@GST_API_VERSION@include_HEADERS)'; test -n "$(libgstrtspserver_@GST_API_VERSION@includedir)" || list=; \
+ files=`for p in $$list; do echo $$p; done | sed -e 's|^.*/||'`; \
+ dir='$(DESTDIR)$(libgstrtspserver_@GST_API_VERSION@includedir)'; $(am__uninstall_files_from_dir)
+tags TAGS:
+
+ctags CTAGS:
+
+cscope cscopelist:
+
+
+distdir: $(DISTFILES)
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+ list='$(DISTFILES)'; \
+ dist_files=`for file in $$list; do echo $$file; done | \
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+ case $$dist_files in \
+ */*) $(MKDIR_P) `echo "$$dist_files" | \
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+ sort -u` ;; \
+ esac; \
+ for file in $$dist_files; do \
+ if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+ if test -d $$d/$$file; then \
+ dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+ if test -d "$(distdir)/$$file"; then \
+ find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+ fi; \
+ if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+ cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+ find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+ fi; \
+ cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+ else \
+ test -f "$(distdir)/$$file" \
+ || cp -p $$d/$$file "$(distdir)/$$file" \
+ || exit 1; \
+ fi; \
+ done
+check-am: all-am
+check: check-am
+all-am: Makefile $(LTLIBRARIES) $(DATA) $(HEADERS)
+installdirs:
+ for dir in "$(DESTDIR)$(libdir)" "$(DESTDIR)$(girdir)" "$(DESTDIR)$(typelibsdir)" "$(DESTDIR)$(libgstrtspserver_@GST_API_VERSION@includedir)"; do \
+ test -z "$$dir" || $(MKDIR_P) "$$dir"; \
+ done
+install: install-am
+install-exec: install-exec-am
+install-data: install-data-am
+uninstall: uninstall-am
+
+install-am: all-am
+ @$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-am
+install-strip:
+ if test -z '$(STRIP)'; then \
+ $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+ install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+ install; \
+ else \
+ $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+ install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+ "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+ fi
+mostlyclean-generic:
+
+clean-generic:
+ -test -z "$(CLEANFILES)" || rm -f $(CLEANFILES)
+
+distclean-generic:
+ -test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+ -test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+ @echo "This command is intended for maintainers to use"
+ @echo "it deletes files that may require special tools to rebuild."
+clean: clean-am
+
+clean-am: clean-generic clean-libLTLIBRARIES clean-libtool \
+ mostlyclean-am
+
+distclean: distclean-am
+ -rm -rf ./$(DEPDIR)
+ -rm -f Makefile
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+
+dvi: dvi-am
+
+dvi-am:
+
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+
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+install-data-am: install-girDATA \
+ install-libgstrtspserver_@GST_API_VERSION@includeHEADERS \
+ install-typelibsDATA
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+
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+ -rm -rf ./$(DEPDIR)
+ -rm -f Makefile
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+ mostlyclean-libtool
+
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+
+.MAKE: install-am install-strip
+
+.PHONY: all all-am check check-am clean clean-generic \
+ clean-libLTLIBRARIES clean-libtool cscopelist-am ctags-am \
+ distclean distclean-compile distclean-generic \
+ distclean-libtool distdir dvi dvi-am html html-am info info-am \
+ install install-am install-data install-data-am install-dvi \
+ install-dvi-am install-exec install-exec-am install-girDATA \
+ install-html install-html-am install-info install-info-am \
+ install-libLTLIBRARIES \
+ install-libgstrtspserver_@GST_API_VERSION@includeHEADERS \
+ install-man install-pdf install-pdf-am install-ps \
+ install-ps-am install-strip install-typelibsDATA installcheck \
+ installcheck-am installdirs maintainer-clean \
+ maintainer-clean-generic mostlyclean mostlyclean-compile \
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+ tags-am uninstall uninstall-am uninstall-girDATA \
+ uninstall-libLTLIBRARIES \
+ uninstall-libgstrtspserver_@GST_API_VERSION@includeHEADERS \
+ uninstall-typelibsDATA
+
+
+@HAVE_INTROSPECTION_TRUE@GstRtspServer-@GST_API_VERSION@.gir: $(INTROSPECTION_SCANNER) libgstrtspserver-@GST_API_VERSION@.la
+@HAVE_INTROSPECTION_TRUE@ $(AM_V_GEN)PKG_CONFIG_PATH="$(GST_PKG_CONFIG_PATH)" \
+@HAVE_INTROSPECTION_TRUE@ $(INTROSPECTION_SCANNER) -v --namespace GstRtspServer \
+@HAVE_INTROSPECTION_TRUE@ --nsversion=@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@ --strip-prefix=Gst \
+@HAVE_INTROSPECTION_TRUE@ --warn-all \
+@HAVE_INTROSPECTION_TRUE@ -I$(top_srcdir) \
+@HAVE_INTROSPECTION_TRUE@ -I$(top_builddir) \
+@HAVE_INTROSPECTION_TRUE@ -DIN_GOBJECT_INTROSPECTION=1 \
+@HAVE_INTROSPECTION_TRUE@ --c-include='gst/rtsp-server/rtsp-server.h' \
+@HAVE_INTROSPECTION_TRUE@ --add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@ --add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@ --add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@ --add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-net-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@ --library=libgstrtspserver-@GST_API_VERSION@.la \
+@HAVE_INTROSPECTION_TRUE@ --include=Gst-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@ --include=GstRtsp-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@ --include=GstNet-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@ --libtool="$(top_builddir)/libtool" \
+@HAVE_INTROSPECTION_TRUE@ --pkg gstreamer-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@ --pkg gstreamer-rtsp-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@ --pkg gstreamer-net-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@ --pkg-export gstreamer-rtsp-server-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@ --output $@ \
+@HAVE_INTROSPECTION_TRUE@ $(gir_headers) \
+@HAVE_INTROSPECTION_TRUE@ $(gir_sources)
+
+@HAVE_INTROSPECTION_TRUE@%.typelib: %.gir $(INTROSPECTION_COMPILER)
+@HAVE_INTROSPECTION_TRUE@ $(AM_V_GEN)PKG_CONFIG_PATH="$(GST_PKG_CONFIG_PATH)" \
+@HAVE_INTROSPECTION_TRUE@ $(INTROSPECTION_COMPILER) \
+@HAVE_INTROSPECTION_TRUE@ --includedir=$(srcdir) \
+@HAVE_INTROSPECTION_TRUE@ --includedir=$(builddir) \
+@HAVE_INTROSPECTION_TRUE@ --includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@ --includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@ --includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@ --includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-net-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@ $(INTROSPECTION_COMPILER_OPTS) $< -o $(@F)
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+/**
+ * SECTION:gstwfdmessage
+ * @short_description: Helper methods for dealing with WFD messages
+ *
+ * <refsect2>
+ * <para>
+ * The GstWFDMessage helper functions makes it easy to parse and create WFD
+ * messages.
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <gio/gio.h>
+
+#include "gstwfdmessage.h"
+
+#define EDID_BLOCK_SIZE 128
+
+#define FREE_STRING(field) g_free (field); (field) = NULL
+#define REPLACE_STRING(field, val) FREE_STRING(field); (field) = g_strdup (val)
+
+#define INIT_ARRAY(field, type, init_func) \
+G_STMT_START { \
+ if (field) { \
+ guint i; \
+ for(i = 0; i < (field)->len; i++) \
+ init_func (&g_array_index ((field), type, i)); \
+ g_array_set_size ((field), 0); \
+ } \
+ else \
+ (field) = g_array_new (FALSE, TRUE, sizeof (type)); \
+} G_STMT_END
+
+#define FREE_ARRAY(field) \
+G_STMT_START { \
+ if (field) \
+ g_array_free ((field), TRUE); \
+ (field) = NULL; \
+} G_STMT_END
+
+#define DEFINE_STRING_SETTER(field) \
+GstWFDResult gst_wfd_message_set_##field (GstWFDMessage *msg, const gchar *val) { \
+ g_free (msg->field); \
+ msg->field = g_strdup (val); \
+ return GST_WFD_OK; \
+}
+#define DEFINE_STRING_GETTER(field) \
+const gchar* gst_wfd_message_get_##field (const GstWFDMessage *msg) { \
+ return msg->field; \
+}
+
+#define DEFINE_ARRAY_LEN(field) \
+guint gst_wfd_message_##field##_len (const GstWFDMessage *msg) { \
+ return msg->field->len; \
+}
+#define DEFINE_ARRAY_GETTER(method, field, type) \
+const type * gst_wfd_message_get_##method (const GstWFDMessage *msg, guint idx) { \
+ return &g_array_index (msg->field, type, idx); \
+}
+#define DEFINE_PTR_ARRAY_GETTER(method, field, type) \
+const type gst_wfd_message_get_##method (const GstWFDMessage *msg, guint idx) { \
+ return g_array_index (msg->field, type, idx); \
+}
+#define DEFINE_ARRAY_INSERT(method, field, intype, dup_method, type) \
+GstWFDResult gst_wfd_message_insert_##method (GstWFDMessage *msg, gint idx, intype val) { \
+ type vt; \
+ type* v = &vt; \
+ dup_method (v, val); \
+ if (idx == -1) \
+ g_array_append_val (msg->field, vt); \
+ else \
+ g_array_insert_val (msg->field, idx, vt); \
+ return GST_WFD_OK; \
+}
+
+#define DEFINE_ARRAY_REPLACE(method, field, intype, free_method, dup_method, type) \
+GstWFDResult gst_wfd_message_replace_##method (GstWFDMessage *msg, guint idx, intype val) { \
+ type *v = &g_array_index (msg->field, type, idx); \
+ free_method (v); \
+ dup_method (v, val); \
+ return GST_WFD_OK; \
+}
+#define DEFINE_ARRAY_REMOVE(method, field, type, free_method) \
+GstWFDResult gst_wfd_message_remove_##method (GstWFDMessage *msg, guint idx) { \
+ type *v = &g_array_index (msg->field, type, idx); \
+ free_method (v); \
+ g_array_remove_index (msg->field, idx); \
+ return GST_WFD_OK; \
+}
+#define DEFINE_ARRAY_ADDER(method, type) \
+GstWFDResult gst_wfd_message_add_##method (GstWFDMessage *msg, const type val) { \
+ return gst_wfd_message_insert_##method (msg, -1, val); \
+}
+
+#define dup_string(v,val) ((*v) = g_strdup (val))
+#define INIT_STR_ARRAY(field) \
+ INIT_ARRAY (field, gchar *, free_string)
+#define DEFINE_STR_ARRAY_GETTER(method, field) \
+ DEFINE_PTR_ARRAY_GETTER(method, field, gchar *)
+#define DEFINE_STR_ARRAY_INSERT(method, field) \
+ DEFINE_ARRAY_INSERT (method, field, const gchar *, dup_string, gchar *)
+#define DEFINE_STR_ARRAY_ADDER(method, field) \
+ DEFINE_ARRAY_ADDER (method, gchar *)
+#define DEFINE_STR_ARRAY_REPLACE(method, field) \
+ DEFINE_ARRAY_REPLACE (method, field, const gchar *, free_string, dup_string, gchar *)
+#define DEFINE_STR_ARRAY_REMOVE(method, field) \
+ DEFINE_ARRAY_REMOVE (method, field, gchar *, free_string)
+
+static GstWFDMessage *gst_wfd_message_boxed_copy (GstWFDMessage * orig);
+static void gst_wfd_message_boxed_free (GstWFDMessage * msg);
+
+G_DEFINE_BOXED_TYPE (GstWFDMessage, gst_wfd_message, gst_wfd_message_boxed_copy,
+ gst_wfd_message_boxed_free);
+
+static GstWFDMessage *
+gst_wfd_message_boxed_copy (GstWFDMessage * orig)
+{
+ GstWFDMessage *copy;
+
+ if (gst_wfd_message_copy (orig, ©) == GST_WFD_OK)
+ return copy;
+
+ return NULL;
+}
+
+static void
+gst_wfd_message_boxed_free (GstWFDMessage * msg)
+{
+ gst_wfd_message_free (msg);
+}
+
+/**
+ * gst_wfd_message_new:
+ * @msg: (out) (transfer full): pointer to new #GstWFDMessage
+ *
+ * Allocate a new GstWFDMessage and store the result in @msg.
+ *
+ * Returns: a #GstWFDResult.
+ */
+GstWFDResult
+gst_wfd_message_new (GstWFDMessage ** msg)
+{
+ GstWFDMessage *newmsg;
+
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ newmsg = g_new0 (GstWFDMessage, 1);
+
+ *msg = newmsg;
+
+ return gst_wfd_message_init (newmsg);
+}
+
+/**
+ * gst_wfd_message_init:
+ * @msg: a #GstWFDMessage
+ *
+ * Initialize @msg so that its contents are as if it was freshly allocated
+ * with gst_wfd_message_new(). This function is mostly used to initialize a message
+ * allocated on the stack. gst_wfd_message_uninit() undoes this operation.
+ *
+ * When this function is invoked on newly allocated data (with malloc or on the
+ * stack), its contents should be set to 0 before calling this function.
+ *
+ * Returns: a #GstWFDResult.
+ */
+GstWFDResult
+gst_wfd_message_init (GstWFDMessage * msg)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ return GST_WFD_OK;
+}
+
+/**
+ * gst_wfd_message_uninit:
+ * @msg: a #GstWFDMessage
+ *
+ * Free all resources allocated in @msg. @msg should not be used anymore after
+ * this function. This function should be used when @msg was allocated on the
+ * stack and initialized with gst_wfd_message_init().
+ *
+ * Returns: a #GstWFDResult.
+ */
+GstWFDResult
+gst_wfd_message_uninit (GstWFDMessage * msg)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (msg->audio_codecs) {
+ guint i = 0;
+ if (msg->audio_codecs->list) {
+ for (; i < msg->audio_codecs->count; i++) {
+ FREE_STRING(msg->audio_codecs->list[i].audio_format);
+ msg->audio_codecs->list[i].modes = 0;
+ msg->audio_codecs->list[i].latency = 0;
+ }
+ FREE_STRING(msg->audio_codecs->list);
+ }
+ FREE_STRING(msg->audio_codecs);
+ }
+
+ if (msg->video_formats) {
+ FREE_STRING(msg->video_formats->list);
+ FREE_STRING(msg->video_formats);
+ }
+
+ if (msg->video_3d_formats) {
+ FREE_STRING(msg->video_3d_formats->list);
+ FREE_STRING(msg->video_3d_formats);
+ }
+
+ if (msg->content_protection) {
+ if (msg->content_protection->hdcp2_spec) {
+ FREE_STRING(msg->content_protection->hdcp2_spec->hdcpversion);
+ FREE_STRING(msg->content_protection->hdcp2_spec->TCPPort);
+ FREE_STRING(msg->content_protection->hdcp2_spec);
+ }
+ FREE_STRING(msg->content_protection);
+ }
+
+ if (msg->display_edid) {
+ if (msg->display_edid->edid_payload)
+ FREE_STRING(msg->display_edid->edid_payload);
+ FREE_STRING(msg->display_edid);
+ }
+
+ if (msg->coupled_sink) {
+ if (msg->coupled_sink->coupled_sink_cap) {
+ FREE_STRING(msg->coupled_sink->coupled_sink_cap->sink_address);
+ FREE_STRING(msg->coupled_sink->coupled_sink_cap);
+ }
+ FREE_STRING(msg->coupled_sink);
+ }
+
+ if (msg->trigger_method) {
+ FREE_STRING(msg->trigger_method->wfd_trigger_method);
+ FREE_STRING(msg->trigger_method);
+ }
+
+ if (msg->presentation_url) {
+ FREE_STRING(msg->presentation_url->wfd_url0);
+ FREE_STRING(msg->presentation_url->wfd_url1);
+ FREE_STRING(msg->presentation_url);
+ }
+
+ if (msg->client_rtp_ports) {
+ FREE_STRING(msg->client_rtp_ports->profile);
+ FREE_STRING(msg->client_rtp_ports->mode);
+ FREE_STRING(msg->client_rtp_ports);
+ }
+
+ if (msg->route) {
+ FREE_STRING(msg->route->destination);
+ FREE_STRING(msg->route);
+ }
+
+ if (msg->I2C) {
+ FREE_STRING(msg->I2C);
+ }
+
+ if (msg->av_format_change_timing) {
+ FREE_STRING(msg->av_format_change_timing);
+ }
+
+ if (msg->preferred_display_mode) {
+ FREE_STRING(msg->preferred_display_mode);
+ }
+
+ if (msg->standby_resume_capability) {
+ FREE_STRING(msg->standby_resume_capability);
+ }
+
+ if (msg->standby) {
+ FREE_STRING(msg->standby);
+ }
+
+ if (msg->connector_type) {
+ FREE_STRING(msg->connector_type);
+ }
+
+ if (msg->idr_request) {
+ FREE_STRING(msg->idr_request);
+ }
+
+ return GST_WFD_OK;
+}
+
+/**
+ * gst_wfd_message_copy:
+ * @msg: a #GstWFDMessage
+ * @copy: (out) (transfer full): pointer to new #GstWFDMessage
+ *
+ * Allocate a new copy of @msg and store the result in @copy. The value in
+ * @copy should be release with gst_wfd_message_free function.
+ *
+ * Returns: a #GstWFDResult
+ *
+ * Since: 1.6
+ */
+GstWFDResult
+gst_wfd_message_copy (const GstWFDMessage * msg, GstWFDMessage ** copy)
+{
+ GstWFDResult ret;
+ GstWFDMessage *cp;
+
+ if (msg == NULL)
+ return GST_WFD_EINVAL;
+
+ ret = gst_wfd_message_new (copy);
+ if (ret != GST_WFD_OK)
+ return ret;
+
+ cp = *copy;
+
+ /* TODO-WFD */
+ if (msg->client_rtp_ports) {
+ cp->client_rtp_ports = g_malloc (sizeof (GstWFDClientRtpPorts));
+ if (cp->client_rtp_ports) {
+ cp->client_rtp_ports->profile = g_strdup (msg->client_rtp_ports->profile);
+ cp->client_rtp_ports->rtp_port0 = msg->client_rtp_ports->rtp_port0;
+ cp->client_rtp_ports->rtp_port1 = msg->client_rtp_ports->rtp_port1;
+ cp->client_rtp_ports->mode = g_strdup (msg->client_rtp_ports->mode);
+ }
+ }
+
+ return GST_WFD_OK;
+}
+
+
+static void
+_read_string_space_ended (gchar * dest, guint size, gchar * src)
+{
+ guint idx = 0;
+
+ while (!g_ascii_isspace (*src) && *src != '\0') {
+ if (idx < size - 1)
+ dest[idx++] = *src;
+ src++;
+ }
+
+ if (size > 0)
+ dest[idx] = '\0';
+
+ return;
+}
+
+static void
+_read_string_attr_and_value (gchar * attr, gchar * value, guint tsize,
+ guint vsize, gchar del, gchar * src)
+{
+ guint idx;
+
+ idx = 0;
+
+ while (*src != del && *src != '\0') {
+ if (idx < tsize - 1)
+ attr[idx++] = *src;
+ src++;
+ }
+
+ if (tsize > 0)
+ attr[idx] = '\0';
+
+ src++;
+ idx = 0;
+
+ while (*src != '\0') {
+ if (idx < vsize - 1)
+ value[idx++] = *src;
+ src++;
+ }
+
+ if (vsize > 0)
+ value[idx] = '\0';
+
+ return;
+}
+
+static void
+gst_wfd_parse_attribute (gchar * buffer, GstWFDMessage * msg)
+{
+ gchar attr[8192] = { 0 };
+ gchar value[8192] = { 0 };
+ gchar temp[8192] = { 0 };
+ gchar *p = buffer;
+ gchar *v = value;
+
+#define WFD_SKIP_SPACE(q) if (*q && g_ascii_isspace (*q)) q++
+#define WFD_SKIP_EQUAL(q) if (*q && *q == '=') q++
+#define WFD_SKIP_COMMA(q) if (*q && g_ascii_ispunct (*q)) q++
+#define WFD_READ_STRING(field) _read_string_space_ended (temp, sizeof (temp), v); v+=strlen(temp); REPLACE_STRING (field, temp)
+#define WFD_READ_UINT32(field) _read_string_space_ended (temp, sizeof (temp), v); v+=strlen(temp); field = strtoul (temp, NULL, 16)
+#define WFD_READ_UINT32_DIGIT(field) _read_string_space_ended (temp, sizeof (temp), v); v+=strlen(temp); field = strtoul (temp, NULL, 10)
+
+ _read_string_attr_and_value (attr, value, sizeof (attr), sizeof (value), ':',
+ p);
+
+ if (!g_strcmp0 (attr, GST_STRING_WFD_AUDIO_CODECS)) {
+ msg->audio_codecs = g_new0 (GstWFDAudioCodeclist, 1);
+ if (strlen (v)) {
+ guint i = 0;
+ msg->audio_codecs->count = strlen (v) / 16;
+ msg->audio_codecs->list =
+ g_new0 (GstWFDAudioCodec, msg->audio_codecs->count);
+ for (; i < msg->audio_codecs->count; i++) {
+ WFD_SKIP_SPACE (v);
+ WFD_READ_STRING (msg->audio_codecs->list[i].audio_format);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->audio_codecs->list[i].modes);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->audio_codecs->list[i].latency);
+ WFD_SKIP_COMMA (v);
+ }
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_VIDEO_FORMATS)) {
+ msg->video_formats = g_new0 (GstWFDVideoCodeclist, 1);
+ if (strlen (v)) {
+ msg->video_formats->count = 1;
+ msg->video_formats->list = g_new0 (GstWFDVideoCodec, 1);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->native);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->
+ list->preferred_display_mode_supported);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.profile);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.level);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+ misc_params.CEA_Support);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+ misc_params.VESA_Support);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+ misc_params.HH_Support);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+ misc_params.latency);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+ misc_params.min_slice_size);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+ misc_params.slice_enc_params);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+ misc_params.frame_rate_control_support);
+ WFD_SKIP_SPACE (v);
+ if (msg->video_formats->list->preferred_display_mode_supported == 1) {
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.max_hres);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.max_vres);
+ WFD_SKIP_SPACE (v);
+ }
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_3D_VIDEO_FORMATS)) {
+ msg->video_3d_formats = g_new0 (GstWFD3DFormats, 1);
+ if (strlen (v)) {
+ msg->video_3d_formats->count = 1;
+ msg->video_3d_formats->list = g_new0 (GstWFD3dCapList, 1);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_3d_formats->list->native);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_3d_formats->
+ list->preferred_display_mode_supported);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.profile);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.level);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.
+ misc_params.video_3d_capability);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.
+ misc_params.latency);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.
+ misc_params.min_slice_size);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.
+ misc_params.slice_enc_params);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.
+ misc_params.frame_rate_control_support);
+ WFD_SKIP_SPACE (v);
+ if (msg->video_formats->list->preferred_display_mode_supported == 1) {
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.max_hres);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->video_formats->list->H264_codec.max_vres);
+ WFD_SKIP_SPACE (v);
+ }
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_CONTENT_PROTECTION)) {
+ msg->content_protection = g_new0 (GstWFDContentProtection, 1);
+ if (strlen (v)) {
+ WFD_SKIP_SPACE (v);
+ msg->content_protection->hdcp2_spec = g_new0 (GstWFDHdcp2Spec, 1);
+ if (strstr (v, "none")) {
+ msg->content_protection->hdcp2_spec->hdcpversion = g_strdup ("none");
+ } else {
+ WFD_READ_STRING (msg->content_protection->hdcp2_spec->hdcpversion);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_STRING (msg->content_protection->hdcp2_spec->TCPPort);
+ }
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_DISPLAY_EDID)) {
+ msg->display_edid = g_new0 (GstWFDDisplayEdid, 1);
+ if (strlen (v)) {
+ WFD_SKIP_SPACE (v);
+ if (strstr (v, "none")) {
+ msg->display_edid->edid_supported = 0;
+ } else {
+ msg->display_edid->edid_supported = 1;
+ WFD_READ_UINT32 (msg->display_edid->edid_block_count);
+ WFD_SKIP_SPACE (v);
+ if (msg->display_edid->edid_block_count) {
+ gchar *edid_string = v;
+ int i = 0, j = 0;
+ guint32 payload_size =
+ EDID_BLOCK_SIZE * msg->display_edid->edid_block_count;
+ msg->display_edid->edid_payload = g_malloc (payload_size);
+ for (;
+ i < (EDID_BLOCK_SIZE * msg->display_edid->edid_block_count * 2);
+ j++) {
+ int k = 0, kk = 0;
+ if (edid_string[i] > 0x29 && edid_string[i] < 0x40)
+ k = edid_string[i] - 48;
+ else if (edid_string[i] > 0x60 && edid_string[i] < 0x67)
+ k = edid_string[i] - 87;
+ else if (edid_string[i] > 0x40 && edid_string[i] < 0x47)
+ k = edid_string[i] - 55;
+
+ if (edid_string[i + 1] > 0x29 && edid_string[i + 1] < 0x40)
+ kk = edid_string[i + 1] - 48;
+ else if (edid_string[i + 1] > 0x60 && edid_string[i + 1] < 0x67)
+ kk = edid_string[i + 1] - 87;
+ else if (edid_string[i + 1] > 0x40 && edid_string[i + 1] < 0x47)
+ kk = edid_string[i + 1] - 55;
+
+ msg->display_edid->edid_payload[j] = (k << 4) | kk;
+ i += 2;
+ }
+ //memcpy(msg->display_edid->edid_payload, v, payload_size);
+ v += (payload_size * 2);
+ } else
+ v += strlen (v);
+ }
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_COUPLED_SINK)) {
+ msg->coupled_sink = g_new0 (GstWFDCoupledSink, 1);
+ if (strlen (v)) {
+ msg->coupled_sink->coupled_sink_cap = g_new0 (GstWFDCoupled_sink_cap, 1);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->coupled_sink->coupled_sink_cap->status);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_STRING (msg->coupled_sink->coupled_sink_cap->sink_address);
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_TRIGGER_METHOD)) {
+ msg->trigger_method = g_new0 (GstWFDTriggerMethod, 1);
+ if (strlen (v)) {
+ WFD_SKIP_SPACE (v);
+ WFD_READ_STRING (msg->trigger_method->wfd_trigger_method);
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_PRESENTATION_URL)) {
+ msg->presentation_url = g_new0 (GstWFDPresentationUrl, 1);
+ if (strlen (v)) {
+ WFD_SKIP_SPACE (v);
+ WFD_READ_STRING (msg->presentation_url->wfd_url0);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_STRING (msg->presentation_url->wfd_url1);
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_CLIENT_RTP_PORTS)) {
+ msg->client_rtp_ports = g_new0 (GstWFDClientRtpPorts, 1);
+ if (strlen (v)) {
+ WFD_SKIP_SPACE (v);
+ WFD_READ_STRING (msg->client_rtp_ports->profile);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32_DIGIT (msg->client_rtp_ports->rtp_port0);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32_DIGIT (msg->client_rtp_ports->rtp_port1);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_STRING (msg->client_rtp_ports->mode);
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_ROUTE)) {
+ msg->route = g_new0 (GstWFDRoute, 1);
+ if (strlen (v)) {
+ WFD_SKIP_SPACE (v);
+ WFD_READ_STRING (msg->route->destination);
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_I2C)) {
+ msg->I2C = g_new0 (GstWFDI2C, 1);
+ if (strlen (v)) {
+ msg->I2C->I2CPresent = TRUE;
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32_DIGIT (msg->I2C->I2C_port);
+ if (msg->I2C->I2C_port)
+ msg->I2C->I2CPresent = TRUE;
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_AV_FORMAT_CHANGE_TIMING)) {
+ msg->av_format_change_timing = g_new0 (GstWFDAVFormatChangeTiming, 1);
+ if (strlen (v)) {
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->av_format_change_timing->PTS);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->av_format_change_timing->DTS);
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_PREFERRED_DISPLAY_MODE)) {
+ msg->preferred_display_mode = g_new0 (GstWFDPreferredDisplayMode, 1);
+ if (strlen (v)) {
+ WFD_SKIP_SPACE (v);
+ if (!strstr (v, "none")) {
+ msg->preferred_display_mode->displaymodesupported = FALSE;
+ } else {
+ WFD_READ_UINT32 (msg->preferred_display_mode->p_clock);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->HB);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->HSPOL_HSOFF);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->HSW);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->V);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->VB);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->VSPOL_VSOFF);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->VSW);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->VBS3D);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->V2d_s3d_modes);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->P_depth);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.profile);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.level);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+ misc_params.CEA_Support);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+ misc_params.VESA_Support);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+ misc_params.HH_Support);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+ misc_params.latency);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+ misc_params.min_slice_size);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+ misc_params.slice_enc_params);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+ misc_params.frame_rate_control_support);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.max_hres);
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.max_vres);
+ WFD_SKIP_SPACE (v);
+ }
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_STANDBY_RESUME_CAPABILITY)) {
+ msg->standby_resume_capability = g_new0 (GstWFDStandbyResumeCapability, 1);
+ if (strlen (v)) {
+ WFD_SKIP_SPACE (v);
+ if (!g_strcmp0 (v, "supported"))
+ msg->standby_resume_capability->standby_resume_cap = TRUE;
+ else
+ msg->standby_resume_capability->standby_resume_cap = FALSE;
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_STANDBY)) {
+ msg->standby = g_new0 (GstWFDStandby, 1);
+ msg->standby->wfd_standby = TRUE;
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_CONNECTOR_TYPE)) {
+ msg->connector_type = g_new0 (GstWFDConnectorType, 1);
+ if (strlen (v)) {
+ msg->connector_type->supported = TRUE;
+ WFD_SKIP_SPACE (v);
+ WFD_READ_UINT32 (msg->connector_type->connector_type);
+ }
+ } else if (!g_strcmp0 (attr, GST_STRING_WFD_IDR_REQUEST)) {
+ msg->idr_request = g_new0 (GstWFDIdrRequest, 1);
+ msg->idr_request->idr_request = TRUE;
+ }
+ return;
+}
+
+/**
+ * gst_wfd_message_parse_buffer:
+ * @data: the start of the buffer
+ * @size: the size of the buffer
+ * @msg: the result #GstSDPMessage
+ *
+ * Parse the contents of @size bytes pointed to by @data and store the result in
+ * @msg.
+ *
+ * Returns: #GST_SDP_OK on success.
+ */
+GstWFDResult
+gst_wfd_message_parse_buffer (const guint8 * data, guint size,
+ GstWFDMessage * msg)
+{
+ gchar *p;
+ gchar buffer[255] = { 0 };
+ guint idx = 0;
+
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+ g_return_val_if_fail (data != NULL, GST_WFD_EINVAL);
+ g_return_val_if_fail (size != 0, GST_WFD_EINVAL);
+
+ p = (gchar *) data;
+ while (TRUE) {
+
+ if (*p == '\0')
+ break;
+
+ idx = 0;
+ while (*p != '\n' && *p != '\r' && *p != '\0') {
+ if (idx < sizeof (buffer) - 1)
+ buffer[idx++] = *p;
+ p++;
+ }
+ buffer[idx] = '\0';
+ gst_wfd_parse_attribute (buffer, msg);
+
+ if (*p == '\0')
+ break;
+ p += 2;
+ }
+ return GST_WFD_OK;
+}
+
+/**
+ * gst_wfd_message_free:
+ * @msg: a #GstWFDMessage
+ *
+ * Free all resources allocated by @msg. @msg should not be used anymore after
+ * this function. This function should be used when @msg was dynamically
+ * allocated with gst_wfd_message_new().
+ *
+ * Returns: a #GstWFDResult.
+ */
+GstWFDResult
+gst_wfd_message_free (GstWFDMessage * msg)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ gst_wfd_message_uninit (msg);
+ g_free (msg);
+
+ return GST_WFD_OK;
+}
+
+/**
+ * gst_wfd_message_as_text:
+ * @msg: a #GstWFDMessage
+ *
+ * Convert the contents of @msg to a text string.
+ *
+ * Returns: A dynamically allocated string representing the WFD description.
+ */
+gchar *
+gst_wfd_message_as_text (const GstWFDMessage * msg)
+{
+ /* change all vars so they match rfc? */
+ GString *lines;
+ guint i;
+
+ g_return_val_if_fail (msg != NULL, NULL);
+
+ lines = g_string_new ("");
+
+ /* list of audio codecs */
+ if (msg->audio_codecs) {
+ g_string_append_printf (lines, GST_STRING_WFD_AUDIO_CODECS);
+ if (msg->audio_codecs->list) {
+ g_string_append_printf (lines, ":");
+ for (i = 0; i < msg->audio_codecs->count; i++) {
+ g_string_append_printf (lines, " %s",
+ msg->audio_codecs->list[i].audio_format);
+ g_string_append_printf (lines, " %08x",
+ msg->audio_codecs->list[i].modes);
+ g_string_append_printf (lines, " %02x",
+ msg->audio_codecs->list[i].latency);
+ if ((i + 1) < msg->audio_codecs->count)
+ g_string_append_printf (lines, ",");
+ }
+ }
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ /* list of video codecs */
+ if (msg->video_formats) {
+ g_string_append_printf (lines, GST_STRING_WFD_VIDEO_FORMATS);
+ if (msg->video_formats->list) {
+ g_string_append_printf (lines, ":");
+ g_string_append_printf (lines, " %02x", msg->video_formats->list->native);
+ g_string_append_printf (lines, " %02x",
+ msg->video_formats->list->preferred_display_mode_supported);
+ g_string_append_printf (lines, " %02x",
+ msg->video_formats->list->H264_codec.profile);
+ g_string_append_printf (lines, " %02x",
+ msg->video_formats->list->H264_codec.level);
+ g_string_append_printf (lines, " %08x",
+ msg->video_formats->list->H264_codec.misc_params.CEA_Support);
+ g_string_append_printf (lines, " %08x",
+ msg->video_formats->list->H264_codec.misc_params.VESA_Support);
+ g_string_append_printf (lines, " %08x",
+ msg->video_formats->list->H264_codec.misc_params.HH_Support);
+ g_string_append_printf (lines, " %02x",
+ msg->video_formats->list->H264_codec.misc_params.latency);
+ g_string_append_printf (lines, " %04x",
+ msg->video_formats->list->H264_codec.misc_params.min_slice_size);
+ g_string_append_printf (lines, " %04x",
+ msg->video_formats->list->H264_codec.misc_params.slice_enc_params);
+ g_string_append_printf (lines, " %02x",
+ msg->video_formats->list->H264_codec.
+ misc_params.frame_rate_control_support);
+
+ if (msg->video_formats->list->H264_codec.max_hres)
+ g_string_append_printf (lines, " %04x",
+ msg->video_formats->list->H264_codec.max_hres);
+ else
+ g_string_append_printf (lines, " none");
+
+ if (msg->video_formats->list->H264_codec.max_vres)
+ g_string_append_printf (lines, " %04x",
+ msg->video_formats->list->H264_codec.max_vres);
+ else
+ g_string_append_printf (lines, " none");
+ }
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ /* list of video 3D codecs */
+ if (msg->video_3d_formats) {
+ g_string_append_printf (lines, GST_STRING_WFD_3D_VIDEO_FORMATS);
+ g_string_append_printf (lines, ":");
+ if (msg->video_3d_formats->list) {
+ g_string_append_printf (lines, " %02x",
+ msg->video_3d_formats->list->native);
+ g_string_append_printf (lines, " %02x",
+ msg->video_3d_formats->list->preferred_display_mode_supported);
+ g_string_append_printf (lines, " %02x",
+ msg->video_3d_formats->list->H264_codec.profile);
+ g_string_append_printf (lines, " %02x",
+ msg->video_3d_formats->list->H264_codec.level);
+ g_string_append_printf (lines, " %16x",
+ msg->video_3d_formats->list->H264_codec.
+ misc_params.video_3d_capability);
+ g_string_append_printf (lines, " %02x",
+ msg->video_3d_formats->list->H264_codec.misc_params.latency);
+ g_string_append_printf (lines, " %04x",
+ msg->video_3d_formats->list->H264_codec.misc_params.min_slice_size);
+ g_string_append_printf (lines, " %04x",
+ msg->video_3d_formats->list->H264_codec.misc_params.slice_enc_params);
+ g_string_append_printf (lines, " %02x",
+ msg->video_3d_formats->list->H264_codec.
+ misc_params.frame_rate_control_support);
+ if (msg->video_3d_formats->list->H264_codec.max_hres)
+ g_string_append_printf (lines, " %04x",
+ msg->video_3d_formats->list->H264_codec.max_hres);
+ else
+ g_string_append_printf (lines, " none");
+ if (msg->video_3d_formats->list->H264_codec.max_vres)
+ g_string_append_printf (lines, " %04x",
+ msg->video_3d_formats->list->H264_codec.max_vres);
+ else
+ g_string_append_printf (lines, " none");
+ } else {
+ g_string_append_printf (lines, " none");
+ }
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->content_protection) {
+ g_string_append_printf (lines, GST_STRING_WFD_CONTENT_PROTECTION);
+ g_string_append_printf (lines, ":");
+ if (msg->content_protection->hdcp2_spec) {
+ if (msg->content_protection->hdcp2_spec->hdcpversion) {
+ g_string_append_printf (lines, " %s",
+ msg->content_protection->hdcp2_spec->hdcpversion);
+ g_string_append_printf (lines, " %s",
+ msg->content_protection->hdcp2_spec->TCPPort);
+ } else {
+ g_string_append_printf (lines, " none");
+ }
+ } else {
+ g_string_append_printf (lines, " none");
+ }
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->display_edid) {
+ g_string_append_printf (lines, GST_STRING_WFD_DISPLAY_EDID);
+ g_string_append_printf (lines, ":");
+ if (msg->display_edid->edid_supported) {
+ g_string_append_printf (lines, " %d", msg->display_edid->edid_supported);
+ if (msg->display_edid->edid_block_count)
+ g_string_append_printf (lines, " %d",
+ msg->display_edid->edid_block_count);
+ else
+ g_string_append_printf (lines, " none");
+ } else {
+ g_string_append_printf (lines, " none");
+ }
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->coupled_sink) {
+ g_string_append_printf (lines, GST_STRING_WFD_COUPLED_SINK);
+ g_string_append_printf (lines, ":");
+ if (msg->coupled_sink->coupled_sink_cap) {
+ g_string_append_printf (lines, " %02x",
+ msg->coupled_sink->coupled_sink_cap->status);
+ if (msg->coupled_sink->coupled_sink_cap->sink_address)
+ g_string_append_printf (lines, " %s",
+ msg->coupled_sink->coupled_sink_cap->sink_address);
+ else
+ g_string_append_printf (lines, " none");
+ } else {
+ g_string_append_printf (lines, " none");
+ }
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->trigger_method) {
+ g_string_append_printf (lines, GST_STRING_WFD_TRIGGER_METHOD);
+ g_string_append_printf (lines, ":");
+ g_string_append_printf (lines, " %s",
+ msg->trigger_method->wfd_trigger_method);
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->presentation_url) {
+ g_string_append_printf (lines, GST_STRING_WFD_PRESENTATION_URL);
+ g_string_append_printf (lines, ":");
+ if (msg->presentation_url->wfd_url0)
+ g_string_append_printf (lines, " %s", msg->presentation_url->wfd_url0);
+ else
+ g_string_append_printf (lines, " none");
+ if (msg->presentation_url->wfd_url1)
+ g_string_append_printf (lines, " %s", msg->presentation_url->wfd_url1);
+ else
+ g_string_append_printf (lines, " none");
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->client_rtp_ports) {
+ g_string_append_printf (lines, GST_STRING_WFD_CLIENT_RTP_PORTS);
+ if (msg->client_rtp_ports->profile) {
+ g_string_append_printf (lines, ":");
+ g_string_append_printf (lines, " %s", msg->client_rtp_ports->profile);
+ g_string_append_printf (lines, " %d", msg->client_rtp_ports->rtp_port0);
+ g_string_append_printf (lines, " %d", msg->client_rtp_ports->rtp_port1);
+ g_string_append_printf (lines, " %s", msg->client_rtp_ports->mode);
+ }
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->route) {
+ g_string_append_printf (lines, GST_STRING_WFD_ROUTE);
+ g_string_append_printf (lines, ":");
+ g_string_append_printf (lines, " %s", msg->route->destination);
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->I2C) {
+ g_string_append_printf (lines, GST_STRING_WFD_I2C);
+ g_string_append_printf (lines, ":");
+ if (msg->I2C->I2CPresent)
+ g_string_append_printf (lines, " %x", msg->I2C->I2C_port);
+ else
+ g_string_append_printf (lines, " none");
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->av_format_change_timing) {
+ g_string_append_printf (lines, GST_STRING_WFD_AV_FORMAT_CHANGE_TIMING);
+ g_string_append_printf (lines, ":");
+ g_string_append_printf (lines, " %010llx",
+ msg->av_format_change_timing->PTS);
+ g_string_append_printf (lines, " %010llx",
+ msg->av_format_change_timing->DTS);
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->preferred_display_mode) {
+ g_string_append_printf (lines, GST_STRING_WFD_PREFERRED_DISPLAY_MODE);
+ g_string_append_printf (lines, ":");
+ if (msg->preferred_display_mode->displaymodesupported) {
+ g_string_append_printf (lines, " %06llx",
+ msg->preferred_display_mode->p_clock);
+ g_string_append_printf (lines, " %04x", msg->preferred_display_mode->H);
+ g_string_append_printf (lines, " %04x", msg->preferred_display_mode->HB);
+ g_string_append_printf (lines, " %04x",
+ msg->preferred_display_mode->HSPOL_HSOFF);
+ g_string_append_printf (lines, " %04x", msg->preferred_display_mode->HSW);
+ g_string_append_printf (lines, " %04x", msg->preferred_display_mode->V);
+ g_string_append_printf (lines, " %04x", msg->preferred_display_mode->VB);
+ g_string_append_printf (lines, " %04x",
+ msg->preferred_display_mode->VSPOL_VSOFF);
+ g_string_append_printf (lines, " %04x", msg->preferred_display_mode->VSW);
+ g_string_append_printf (lines, " %02x",
+ msg->preferred_display_mode->VBS3D);
+ g_string_append_printf (lines, " %02x",
+ msg->preferred_display_mode->V2d_s3d_modes);
+ g_string_append_printf (lines, " %02x",
+ msg->preferred_display_mode->P_depth);
+ } else
+ g_string_append_printf (lines, " none");
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->standby_resume_capability) {
+ g_string_append_printf (lines, GST_STRING_WFD_STANDBY_RESUME_CAPABILITY);
+ g_string_append_printf (lines, ":");
+ if (msg->standby_resume_capability->standby_resume_cap)
+ g_string_append_printf (lines, " supported");
+ else
+ g_string_append_printf (lines, " none");
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->standby) {
+ g_string_append_printf (lines, GST_STRING_WFD_STANDBY);
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->connector_type) {
+ g_string_append_printf (lines, GST_STRING_WFD_CONNECTOR_TYPE);
+ g_string_append_printf (lines, ":");
+ if (msg->connector_type->connector_type)
+ g_string_append_printf (lines, " %02x",
+ msg->connector_type->connector_type);
+ else
+ g_string_append_printf (lines, " none");
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ if (msg->idr_request) {
+ g_string_append_printf (lines, GST_STRING_WFD_IDR_REQUEST);
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ return g_string_free (lines, FALSE);
+}
+
+gchar *
+gst_wfd_message_param_names_as_text (const GstWFDMessage * msg)
+{
+ /* change all vars so they match rfc? */
+ GString *lines;
+ g_return_val_if_fail (msg != NULL, NULL);
+
+ lines = g_string_new ("");
+
+ /* list of audio codecs */
+ if (msg->audio_codecs) {
+ g_string_append_printf (lines, GST_STRING_WFD_AUDIO_CODECS);
+ g_string_append_printf (lines, "\r\n");
+ }
+ /* list of video codecs */
+ if (msg->video_formats) {
+ g_string_append_printf (lines, GST_STRING_WFD_VIDEO_FORMATS);
+ g_string_append_printf (lines, "\r\n");
+ }
+ /* list of video 3D codecs */
+ if (msg->video_3d_formats) {
+ g_string_append_printf (lines, GST_STRING_WFD_3D_VIDEO_FORMATS);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->content_protection) {
+ g_string_append_printf (lines, GST_STRING_WFD_CONTENT_PROTECTION);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->display_edid) {
+ g_string_append_printf (lines, GST_STRING_WFD_DISPLAY_EDID);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->coupled_sink) {
+ g_string_append_printf (lines, GST_STRING_WFD_COUPLED_SINK);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->trigger_method) {
+ g_string_append_printf (lines, GST_STRING_WFD_TRIGGER_METHOD);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->presentation_url) {
+ g_string_append_printf (lines, GST_STRING_WFD_PRESENTATION_URL);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->client_rtp_ports) {
+ g_string_append_printf (lines, GST_STRING_WFD_CLIENT_RTP_PORTS);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->route) {
+ g_string_append_printf (lines, GST_STRING_WFD_ROUTE);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->I2C) {
+ g_string_append_printf (lines, GST_STRING_WFD_I2C);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->av_format_change_timing) {
+ g_string_append_printf (lines, GST_STRING_WFD_AV_FORMAT_CHANGE_TIMING);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->preferred_display_mode) {
+ g_string_append_printf (lines, GST_STRING_WFD_PREFERRED_DISPLAY_MODE);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->standby_resume_capability) {
+ g_string_append_printf (lines, GST_STRING_WFD_STANDBY_RESUME_CAPABILITY);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->standby) {
+ g_string_append_printf (lines, GST_STRING_WFD_STANDBY);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->connector_type) {
+ g_string_append_printf (lines, GST_STRING_WFD_CONNECTOR_TYPE);
+ g_string_append_printf (lines, "\r\n");
+ }
+ if (msg->idr_request) {
+ g_string_append_printf (lines, GST_STRING_WFD_IDR_REQUEST);
+ g_string_append_printf (lines, "\r\n");
+ }
+
+ return g_string_free (lines, FALSE);
+}
+
+/**
+ * gst_wfd_message_dump:
+ * @msg: a #GstWFDMessage
+ *
+ * Dump the parsed contents of @msg to stdout.
+ *
+ * Returns: a #GstWFDResult.
+ */
+GstWFDResult
+gst_wfd_message_dump (const GstWFDMessage * msg)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (msg->audio_codecs) {
+ guint i = 0;
+ g_print ("Audio supported formats : \n");
+ for (; i < msg->audio_codecs->count; i++) {
+ g_print ("Codec: %s\n", msg->audio_codecs->list[i].audio_format);
+ if (!strcmp (msg->audio_codecs->list[i].audio_format, "LPCM")) {
+ if (msg->audio_codecs->list[i].modes & GST_WFD_FREQ_44100)
+ g_print (" Freq: %d\n", 44100);
+ if (msg->audio_codecs->list[i].modes & GST_WFD_FREQ_48000)
+ g_print (" Freq: %d\n", 48000);
+ g_print (" Channels: %d\n", 2);
+ }
+ if (!strcmp (msg->audio_codecs->list[i].audio_format, "AAC")) {
+ g_print (" Freq: %d\n", 48000);
+ if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_2)
+ g_print (" Channels: %d\n", 2);
+ if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_4)
+ g_print (" Channels: %d\n", 4);
+ if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_6)
+ g_print (" Channels: %d\n", 6);
+ if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_8)
+ g_print (" Channels: %d\n", 8);
+ }
+ if (!strcmp (msg->audio_codecs->list[i].audio_format, "AC3")) {
+ g_print (" Freq: %d\n", 48000);
+ if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_2)
+ g_print (" Channels: %d\n", 2);
+ if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_4)
+ g_print (" Channels: %d\n", 4);
+ if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_6)
+ g_print (" Channels: %d\n", 6);
+ }
+ g_print (" Bitwidth: %d\n", 16);
+ g_print (" Latency: %d\n", msg->audio_codecs->list[i].latency);
+ }
+ }
+
+
+ if (msg->video_formats) {
+ g_print ("Video supported formats : \n");
+ if (msg->video_formats->list) {
+ guint nativeindex = 0;
+ g_print ("Codec: H264\n");
+ if ((msg->video_formats->list->native & 0x7) ==
+ GST_WFD_VIDEO_CEA_RESOLUTION) {
+ g_print (" Native type: CEA\n");
+ } else if ((msg->video_formats->list->native & 0x7) ==
+ GST_WFD_VIDEO_VESA_RESOLUTION) {
+ g_print (" Native type: VESA\n");
+ } else if ((msg->video_formats->list->native & 0x7) ==
+ GST_WFD_VIDEO_HH_RESOLUTION) {
+ g_print (" Native type: HH\n");
+ }
+ nativeindex = msg->video_formats->list->native >> 3;
+ g_print (" Resolution: %d\n", (1 << nativeindex));
+
+ if (msg->video_formats->list->
+ H264_codec.profile & GST_WFD_H264_BASE_PROFILE) {
+ g_print (" Profile: BASE\n");
+ } else if (msg->video_formats->list->
+ H264_codec.profile & GST_WFD_H264_HIGH_PROFILE) {
+ g_print (" Profile: HIGH\n");
+ }
+ if (msg->video_formats->list->H264_codec.level & GST_WFD_H264_LEVEL_3_1) {
+ g_print (" Level: 3.1\n");
+ } else if (msg->video_formats->list->
+ H264_codec.level & GST_WFD_H264_LEVEL_3_2) {
+ g_print (" Level: 3.2\n");
+ } else if (msg->video_formats->list->
+ H264_codec.level & GST_WFD_H264_LEVEL_4) {
+ g_print (" Level: 4\n");
+ } else if (msg->video_formats->list->
+ H264_codec.level & GST_WFD_H264_LEVEL_4_1) {
+ g_print (" Level: 4.1\n");
+ } else if (msg->video_formats->list->
+ H264_codec.level & GST_WFD_H264_LEVEL_4_2) {
+ g_print (" Level: 4.2\n");
+ }
+ g_print (" Latency: %d\n",
+ msg->video_formats->list->H264_codec.misc_params.latency);
+ g_print (" min_slice_size: %x\n",
+ msg->video_formats->list->H264_codec.misc_params.min_slice_size);
+ g_print (" slice_enc_params: %x\n",
+ msg->video_formats->list->H264_codec.misc_params.slice_enc_params);
+ g_print (" frame_rate_control_support: %x\n",
+ msg->video_formats->list->H264_codec.
+ misc_params.frame_rate_control_support);
+ if (msg->video_formats->list->H264_codec.max_hres) {
+ g_print (" Max Height: %04d\n",
+ msg->video_formats->list->H264_codec.max_hres);
+ }
+ if (msg->video_formats->list->H264_codec.max_vres) {
+ g_print (" Max Width: %04d\n",
+ msg->video_formats->list->H264_codec.max_vres);
+ }
+ }
+ }
+
+ if (msg->video_3d_formats) {
+ g_print ("wfd_3d_formats");
+ g_print ("\r\n");
+ }
+
+ if (msg->content_protection) {
+ g_print (GST_STRING_WFD_CONTENT_PROTECTION);
+ g_print ("\r\n");
+ }
+
+ if (msg->display_edid) {
+ g_print (GST_STRING_WFD_DISPLAY_EDID);
+ g_print ("\r\n");
+ }
+
+ if (msg->coupled_sink) {
+ g_print (GST_STRING_WFD_COUPLED_SINK);
+ g_print ("\r\n");
+ }
+
+ if (msg->trigger_method) {
+ g_print (" Trigger type: %s\n", msg->trigger_method->wfd_trigger_method);
+ }
+
+ if (msg->presentation_url) {
+ g_print (GST_STRING_WFD_PRESENTATION_URL);
+ g_print ("\r\n");
+ }
+
+ if (msg->client_rtp_ports) {
+ g_print (" Client RTP Ports : \n");
+ if (msg->client_rtp_ports->profile) {
+ g_print ("%s\n", msg->client_rtp_ports->profile);
+ g_print (" %d\n", msg->client_rtp_ports->rtp_port0);
+ g_print (" %d\n", msg->client_rtp_ports->rtp_port1);
+ g_print (" %s\n", msg->client_rtp_ports->mode);
+ }
+ g_print ("\r\n");
+ }
+
+ if (msg->route) {
+ g_print (GST_STRING_WFD_ROUTE);
+ g_print ("\r\n");
+ }
+
+ if (msg->I2C) {
+ g_print (GST_STRING_WFD_I2C);
+ g_print ("\r\n");
+ }
+
+ if (msg->av_format_change_timing) {
+ g_print (GST_STRING_WFD_AV_FORMAT_CHANGE_TIMING);
+ g_print ("\r\n");
+ }
+
+ if (msg->preferred_display_mode) {
+ g_print (GST_STRING_WFD_PREFERRED_DISPLAY_MODE);
+ g_print ("\r\n");
+ }
+
+ if (msg->standby_resume_capability) {
+ g_print (GST_STRING_WFD_STANDBY_RESUME_CAPABILITY);
+ g_print ("\r\n");
+ }
+
+ if (msg->standby) {
+ g_print (GST_STRING_WFD_STANDBY);
+ g_print ("\r\n");
+ }
+
+ if (msg->connector_type) {
+ g_print (GST_STRING_WFD_CONNECTOR_TYPE);
+ g_print ("\r\n");
+ }
+
+ if (msg->idr_request) {
+ g_print (GST_STRING_WFD_IDR_REQUEST);
+ g_print ("\r\n");
+ }
+
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_supported_audio_format (GstWFDMessage * msg,
+ GstWFDAudioFormats a_codec,
+ guint a_freq, guint a_channels, guint a_bitwidth, guint32 a_latency)
+{
+ guint temp = a_codec;
+ guint i = 0;
+ guint pcm = 0, aac = 0, ac3 = 0;
+
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (!msg->audio_codecs)
+ msg->audio_codecs = g_new0 (GstWFDAudioCodeclist, 1);
+
+ if (a_codec != GST_WFD_AUDIO_UNKNOWN) {
+ while (temp) {
+ msg->audio_codecs->count++;
+ temp >>= 1;
+ }
+ msg->audio_codecs->list =
+ g_new0 (GstWFDAudioCodec, msg->audio_codecs->count);
+ for (; i < msg->audio_codecs->count; i++) {
+ if ((a_codec & GST_WFD_AUDIO_LPCM) && (!pcm)) {
+ msg->audio_codecs->list[i].audio_format = g_strdup ("LPCM");
+ msg->audio_codecs->list[i].modes = a_freq;
+ msg->audio_codecs->list[i].latency = a_latency;
+ pcm = 1;
+ } else if ((a_codec & GST_WFD_AUDIO_AAC) && (!aac)) {
+ msg->audio_codecs->list[i].audio_format = g_strdup ("AAC");
+ msg->audio_codecs->list[i].modes = a_channels;
+ msg->audio_codecs->list[i].latency = a_latency;
+ aac = 1;
+ } else if ((a_codec & GST_WFD_AUDIO_AC3) && (!ac3)) {
+ msg->audio_codecs->list[i].audio_format = g_strdup ("AC3");
+ msg->audio_codecs->list[i].modes = a_channels;
+ msg->audio_codecs->list[i].latency = a_latency;
+ ac3 = 1;
+ }
+ }
+ }
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_prefered_audio_format (GstWFDMessage * msg,
+ GstWFDAudioFormats a_codec,
+ GstWFDAudioFreq a_freq,
+ GstWFDAudioChannels a_channels, guint a_bitwidth, guint32 a_latency)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (!msg->audio_codecs)
+ msg->audio_codecs = g_new0 (GstWFDAudioCodeclist, 1);
+
+ msg->audio_codecs->list = g_new0 (GstWFDAudioCodec, 1);
+ msg->audio_codecs->count = 1;
+ if (a_codec == GST_WFD_AUDIO_LPCM) {
+ msg->audio_codecs->list->audio_format = g_strdup ("LPCM");
+ msg->audio_codecs->list->modes = a_freq;
+ msg->audio_codecs->list->latency = a_latency;
+ } else if (a_codec == GST_WFD_AUDIO_AAC) {
+ msg->audio_codecs->list->audio_format = g_strdup ("AAC");
+ msg->audio_codecs->list->modes = a_channels;
+ msg->audio_codecs->list->latency = a_latency;
+ } else if (a_codec == GST_WFD_AUDIO_AC3) {
+ msg->audio_codecs->list->audio_format = g_strdup ("AC3");
+ msg->audio_codecs->list->modes = a_channels;
+ msg->audio_codecs->list->latency = a_latency;
+ }
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_supported_audio_format (GstWFDMessage * msg,
+ guint * a_codec,
+ guint * a_freq, guint * a_channels, guint * a_bitwidth, guint32 * a_latency)
+{
+ guint i = 0;
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+ g_return_val_if_fail (msg->audio_codecs != NULL, GST_WFD_EINVAL);
+
+ for (; i < msg->audio_codecs->count; i++) {
+ if (!g_strcmp0 (msg->audio_codecs->list[i].audio_format, "LPCM")) {
+ *a_codec |= GST_WFD_AUDIO_LPCM;
+ *a_freq |= msg->audio_codecs->list[i].modes;
+ *a_channels |= GST_WFD_CHANNEL_2;
+ *a_bitwidth = 16;
+ *a_latency = msg->audio_codecs->list[i].latency;
+ } else if (!g_strcmp0 (msg->audio_codecs->list[i].audio_format, "AAC")) {
+ *a_codec |= GST_WFD_AUDIO_AAC;
+ *a_freq |= GST_WFD_FREQ_48000;
+ *a_channels |= msg->audio_codecs->list[i].modes;
+ *a_bitwidth = 16;
+ *a_latency = msg->audio_codecs->list[i].latency;
+ } else if (!g_strcmp0 (msg->audio_codecs->list[i].audio_format, "AC3")) {
+ *a_codec |= GST_WFD_AUDIO_AC3;
+ *a_freq |= GST_WFD_FREQ_48000;
+ *a_channels |= msg->audio_codecs->list[i].modes;
+ *a_bitwidth = 16;
+ *a_latency = msg->audio_codecs->list[i].latency;
+ }
+ }
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_prefered_audio_format (GstWFDMessage * msg,
+ GstWFDAudioFormats * a_codec,
+ GstWFDAudioFreq * a_freq,
+ GstWFDAudioChannels * a_channels, guint * a_bitwidth, guint32 * a_latency)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (!g_strcmp0 (msg->audio_codecs->list->audio_format, "LPCM")) {
+ *a_codec = GST_WFD_AUDIO_LPCM;
+ *a_freq = msg->audio_codecs->list->modes;
+ *a_channels = GST_WFD_CHANNEL_2;
+ *a_bitwidth = 16;
+ *a_latency = msg->audio_codecs->list->latency;
+ } else if (!g_strcmp0 (msg->audio_codecs->list->audio_format, "AAC")) {
+ *a_codec = GST_WFD_AUDIO_AAC;
+ *a_freq = GST_WFD_FREQ_48000;
+ *a_channels = msg->audio_codecs->list->modes;
+ *a_bitwidth = 16;
+ *a_latency = msg->audio_codecs->list->latency;
+ } else if (!g_strcmp0 (msg->audio_codecs->list->audio_format, "AC3")) {
+ *a_codec = GST_WFD_AUDIO_AC3;
+ *a_freq = GST_WFD_FREQ_48000;
+ *a_channels = msg->audio_codecs->list->modes;
+ *a_bitwidth = 16;
+ *a_latency = msg->audio_codecs->list->latency;
+ }
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_supported_video_format (GstWFDMessage * msg,
+ GstWFDVideoCodecs v_codec,
+ GstWFDVideoNativeResolution v_native,
+ guint64 v_native_resolution,
+ guint64 v_cea_resolution,
+ guint64 v_vesa_resolution,
+ guint64 v_hh_resolution,
+ guint v_profile,
+ guint v_level,
+ guint32 v_latency,
+ guint32 v_max_height,
+ guint32 v_max_width,
+ guint32 min_slice_size, guint32 slice_enc_params, guint frame_rate_control)
+{
+ guint nativeindex = 0;
+ guint64 temp = v_native_resolution;
+
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (!msg->video_formats)
+ msg->video_formats = g_new0 (GstWFDVideoCodeclist, 1);
+
+ if (v_codec != GST_WFD_VIDEO_UNKNOWN) {
+ msg->video_formats->list = g_new0 (GstWFDVideoCodec, 1);
+ while (temp) {
+ nativeindex++;
+ temp >>= 1;
+ }
+
+ msg->video_formats->list->native = nativeindex - 1;
+ msg->video_formats->list->native <<= 3;
+
+ if (v_native == GST_WFD_VIDEO_VESA_RESOLUTION)
+ msg->video_formats->list->native |= 1;
+ else if (v_native == GST_WFD_VIDEO_HH_RESOLUTION)
+ msg->video_formats->list->native |= 2;
+
+ msg->video_formats->list->preferred_display_mode_supported = 1;
+ msg->video_formats->list->H264_codec.profile = v_profile;
+ msg->video_formats->list->H264_codec.level = v_level;
+ msg->video_formats->list->H264_codec.max_hres = v_max_height;
+ msg->video_formats->list->H264_codec.max_vres = v_max_width;
+ msg->video_formats->list->H264_codec.misc_params.CEA_Support =
+ v_cea_resolution;
+ msg->video_formats->list->H264_codec.misc_params.VESA_Support =
+ v_vesa_resolution;
+ msg->video_formats->list->H264_codec.misc_params.HH_Support =
+ v_hh_resolution;
+ msg->video_formats->list->H264_codec.misc_params.latency = v_latency;
+ msg->video_formats->list->H264_codec.misc_params.min_slice_size =
+ min_slice_size;
+ msg->video_formats->list->H264_codec.misc_params.slice_enc_params =
+ slice_enc_params;
+ msg->video_formats->list->H264_codec.
+ misc_params.frame_rate_control_support = frame_rate_control;
+ }
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_prefered_video_format (GstWFDMessage * msg,
+ GstWFDVideoCodecs v_codec,
+ GstWFDVideoNativeResolution v_native,
+ guint64 v_native_resolution,
+ GstWFDVideoCEAResolution v_cea_resolution,
+ GstWFDVideoVESAResolution v_vesa_resolution,
+ GstWFDVideoHHResolution v_hh_resolution,
+ GstWFDVideoH264Profile v_profile,
+ GstWFDVideoH264Level v_level,
+ guint32 v_latency,
+ guint32 v_max_height,
+ guint32 v_max_width,
+ guint32 min_slice_size, guint32 slice_enc_params, guint frame_rate_control)
+{
+ guint nativeindex = 0;
+ guint64 temp = v_native_resolution;
+
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (!msg->video_formats)
+ msg->video_formats = g_new0 (GstWFDVideoCodeclist, 1);
+ msg->video_formats->list = g_new0 (GstWFDVideoCodec, 1);
+
+ while (temp) {
+ nativeindex++;
+ temp >>= 1;
+ }
+
+ if (nativeindex)
+ msg->video_formats->list->native = nativeindex - 1;
+ msg->video_formats->list->native <<= 3;
+
+ if (v_native == GST_WFD_VIDEO_VESA_RESOLUTION)
+ msg->video_formats->list->native |= 1;
+ else if (v_native == GST_WFD_VIDEO_HH_RESOLUTION)
+ msg->video_formats->list->native |= 2;
+
+ msg->video_formats->list->preferred_display_mode_supported = 0;
+ msg->video_formats->list->H264_codec.profile = v_profile;
+ msg->video_formats->list->H264_codec.level = v_level;
+ msg->video_formats->list->H264_codec.max_hres = v_max_height;
+ msg->video_formats->list->H264_codec.max_vres = v_max_width;
+ msg->video_formats->list->H264_codec.misc_params.CEA_Support =
+ v_cea_resolution;
+ msg->video_formats->list->H264_codec.misc_params.VESA_Support =
+ v_vesa_resolution;
+ msg->video_formats->list->H264_codec.misc_params.HH_Support = v_hh_resolution;
+ msg->video_formats->list->H264_codec.misc_params.latency = v_latency;
+ msg->video_formats->list->H264_codec.misc_params.min_slice_size =
+ min_slice_size;
+ msg->video_formats->list->H264_codec.misc_params.slice_enc_params =
+ slice_enc_params;
+ msg->video_formats->list->H264_codec.misc_params.frame_rate_control_support =
+ frame_rate_control;
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_supported_video_format (GstWFDMessage * msg,
+ GstWFDVideoCodecs * v_codec,
+ GstWFDVideoNativeResolution * v_native,
+ guint64 * v_native_resolution,
+ guint64 * v_cea_resolution,
+ guint64 * v_vesa_resolution,
+ guint64 * v_hh_resolution,
+ guint * v_profile,
+ guint * v_level,
+ guint32 * v_latency,
+ guint32 * v_max_height,
+ guint32 * v_max_width,
+ guint32 * min_slice_size,
+ guint32 * slice_enc_params, guint * frame_rate_control)
+{
+ guint nativeindex = 0;
+
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+ *v_codec = GST_WFD_VIDEO_H264;
+ *v_native = msg->video_formats->list->native & 0x7;
+ nativeindex = msg->video_formats->list->native >> 3;
+ *v_native_resolution = ((guint64) 1) << nativeindex;
+ *v_profile = msg->video_formats->list->H264_codec.profile;
+ *v_level = msg->video_formats->list->H264_codec.level;
+ *v_max_height = msg->video_formats->list->H264_codec.max_hres;
+ *v_max_width = msg->video_formats->list->H264_codec.max_vres;
+ *v_cea_resolution =
+ msg->video_formats->list->H264_codec.misc_params.CEA_Support;
+ *v_vesa_resolution =
+ msg->video_formats->list->H264_codec.misc_params.VESA_Support;
+ *v_hh_resolution =
+ msg->video_formats->list->H264_codec.misc_params.HH_Support;
+ *v_latency = msg->video_formats->list->H264_codec.misc_params.latency;
+ *min_slice_size =
+ msg->video_formats->list->H264_codec.misc_params.min_slice_size;
+ *slice_enc_params =
+ msg->video_formats->list->H264_codec.misc_params.slice_enc_params;
+ *frame_rate_control =
+ msg->video_formats->list->H264_codec.
+ misc_params.frame_rate_control_support;
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_prefered_video_format (GstWFDMessage * msg,
+ GstWFDVideoCodecs * v_codec,
+ GstWFDVideoNativeResolution * v_native,
+ guint64 * v_native_resolution,
+ GstWFDVideoCEAResolution * v_cea_resolution,
+ GstWFDVideoVESAResolution * v_vesa_resolution,
+ GstWFDVideoHHResolution * v_hh_resolution,
+ GstWFDVideoH264Profile * v_profile,
+ GstWFDVideoH264Level * v_level,
+ guint32 * v_latency,
+ guint32 * v_max_height,
+ guint32 * v_max_width,
+ guint32 * min_slice_size,
+ guint32 * slice_enc_params, guint * frame_rate_control)
+{
+ guint nativeindex = 0;
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ *v_codec = GST_WFD_VIDEO_H264;
+ *v_native = msg->video_formats->list->native & 0x7;
+ nativeindex = msg->video_formats->list->native >> 3;
+ *v_native_resolution = ((guint64) 1) << nativeindex;
+ *v_profile = msg->video_formats->list->H264_codec.profile;
+ *v_level = msg->video_formats->list->H264_codec.level;
+ *v_max_height = msg->video_formats->list->H264_codec.max_hres;
+ *v_max_width = msg->video_formats->list->H264_codec.max_vres;
+ *v_cea_resolution =
+ msg->video_formats->list->H264_codec.misc_params.CEA_Support;
+ *v_vesa_resolution =
+ msg->video_formats->list->H264_codec.misc_params.VESA_Support;
+ *v_hh_resolution =
+ msg->video_formats->list->H264_codec.misc_params.HH_Support;
+ *v_latency = msg->video_formats->list->H264_codec.misc_params.latency;
+ *min_slice_size =
+ msg->video_formats->list->H264_codec.misc_params.min_slice_size;
+ *slice_enc_params =
+ msg->video_formats->list->H264_codec.misc_params.slice_enc_params;
+ *frame_rate_control =
+ msg->video_formats->list->H264_codec.
+ misc_params.frame_rate_control_support;
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_display_edid (GstWFDMessage * msg,
+ gboolean edid_supported, guint32 edid_blockcount, gchar * edid_playload)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+ if (!msg->display_edid)
+ msg->display_edid = g_new0 (GstWFDDisplayEdid, 1);
+ msg->display_edid->edid_supported = edid_supported;
+ if (!edid_supported)
+ return GST_WFD_OK;
+ msg->display_edid->edid_block_count = edid_blockcount;
+ if (edid_blockcount) {
+ msg->display_edid->edid_payload = g_malloc (128 * edid_blockcount);
+ if (!msg->display_edid->edid_payload)
+ memcpy (msg->display_edid->edid_payload, edid_playload,
+ 128 * edid_blockcount);
+ } else
+ msg->display_edid->edid_payload = g_strdup ("none");
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_display_edid (GstWFDMessage * msg,
+ gboolean * edid_supported,
+ guint32 * edid_blockcount, gchar ** edid_playload)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+ if (msg->display_edid) {
+ if (msg->display_edid->edid_supported) {
+ *edid_blockcount = msg->display_edid->edid_block_count;
+ if (msg->display_edid->edid_block_count) {
+ char *temp;
+ temp = g_malloc (EDID_BLOCK_SIZE * msg->display_edid->edid_block_count);
+ if (temp) {
+ memset (temp, 0,
+ EDID_BLOCK_SIZE * msg->display_edid->edid_block_count);
+ memcpy (temp, msg->display_edid->edid_payload,
+ EDID_BLOCK_SIZE * msg->display_edid->edid_block_count);
+ *edid_playload = temp;
+ *edid_supported = TRUE;
+ }
+ } else
+ *edid_playload = g_strdup ("none");
+ }
+ } else
+ *edid_supported = FALSE;
+ return GST_WFD_OK;
+}
+
+
+GstWFDResult
+gst_wfd_message_set_contentprotection_type (GstWFDMessage * msg,
+ GstWFDHDCPProtection hdcpversion, guint32 TCPPort)
+{
+ char str[11] = { 0, };
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (!msg->content_protection)
+ msg->content_protection = g_new0 (GstWFDContentProtection, 1);
+ if (hdcpversion == GST_WFD_HDCP_NONE)
+ return GST_WFD_OK;
+ msg->content_protection->hdcp2_spec = g_new0 (GstWFDHdcp2Spec, 1);
+ if (hdcpversion == GST_WFD_HDCP_2_0)
+ msg->content_protection->hdcp2_spec->hdcpversion = g_strdup ("HDCP2.0");
+ else if (hdcpversion == GST_WFD_HDCP_2_1)
+ msg->content_protection->hdcp2_spec->hdcpversion = g_strdup ("HDCP2.1");
+ snprintf (str, sizeof (str), "port=%d", TCPPort);
+ msg->content_protection->hdcp2_spec->TCPPort = g_strdup (str);
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_contentprotection_type (GstWFDMessage * msg,
+ GstWFDHDCPProtection * hdcpversion, guint32 * TCPPort)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+ if (msg->content_protection && msg->content_protection->hdcp2_spec) {
+ char *result = NULL;
+ char *ptr = NULL;
+ if (!g_strcmp0 (msg->content_protection->hdcp2_spec->hdcpversion, "none")) {
+ *hdcpversion = GST_WFD_HDCP_NONE;
+ *TCPPort = 0;
+ return GST_WFD_OK;
+ }
+ if (!g_strcmp0 (msg->content_protection->hdcp2_spec->hdcpversion,
+ "HDCP2.0"))
+ *hdcpversion = GST_WFD_HDCP_2_0;
+ else if (!g_strcmp0 (msg->content_protection->hdcp2_spec->hdcpversion,
+ "HDCP2.1"))
+ *hdcpversion = GST_WFD_HDCP_2_1;
+ else {
+ *hdcpversion = GST_WFD_HDCP_NONE;
+ *TCPPort = 0;
+ return GST_WFD_OK;
+ }
+
+ result = strtok_r (msg->content_protection->hdcp2_spec->TCPPort, "=", &ptr);
+ while (result != NULL) {
+ result = strtok_r (NULL, "=", &ptr);
+ *TCPPort = atoi (result);
+ break;
+ }
+ } else
+ *hdcpversion = GST_WFD_HDCP_NONE;
+ return GST_WFD_OK;
+}
+
+
+GstWFDResult
+gst_wfd_messge_set_prefered_rtp_ports (GstWFDMessage * msg,
+ GstWFDRTSPTransMode trans,
+ GstWFDRTSPProfile profile,
+ GstWFDRTSPLowerTrans lowertrans, guint32 rtp_port0, guint32 rtp_port1)
+{
+ GString *lines;
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (!msg->client_rtp_ports)
+ msg->client_rtp_ports = g_new0 (GstWFDClientRtpPorts, 1);
+
+ if (trans != GST_WFD_RTSP_TRANS_UNKNOWN) {
+ lines = g_string_new ("");
+ if (trans == GST_WFD_RTSP_TRANS_RTP)
+ g_string_append_printf (lines, "RTP");
+ else if (trans == GST_WFD_RTSP_TRANS_RDT)
+ g_string_append_printf (lines, "RDT");
+
+ if (profile == GST_WFD_RTSP_PROFILE_AVP)
+ g_string_append_printf (lines, "/AVP");
+ else if (profile == GST_WFD_RTSP_PROFILE_SAVP)
+ g_string_append_printf (lines, "/SAVP");
+
+ if (lowertrans == GST_WFD_RTSP_LOWER_TRANS_UDP)
+ g_string_append_printf (lines, "/UDP;unicast");
+ else if (lowertrans == GST_WFD_RTSP_LOWER_TRANS_UDP_MCAST)
+ g_string_append_printf (lines, "/UDP;multicast");
+ else if (lowertrans == GST_WFD_RTSP_LOWER_TRANS_TCP)
+ g_string_append_printf (lines, "/TCP;unicast");
+ else if (lowertrans == GST_WFD_RTSP_LOWER_TRANS_HTTP)
+ g_string_append_printf (lines, "/HTTP");
+
+ msg->client_rtp_ports->profile = g_string_free (lines, FALSE);
+ msg->client_rtp_ports->rtp_port0 = rtp_port0;
+ msg->client_rtp_ports->rtp_port1 = rtp_port1;
+ msg->client_rtp_ports->mode = g_strdup ("mode=play");
+ }
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_prefered_rtp_ports (GstWFDMessage * msg,
+ GstWFDRTSPTransMode * trans,
+ GstWFDRTSPProfile * profile,
+ GstWFDRTSPLowerTrans * lowertrans, guint32 * rtp_port0, guint32 * rtp_port1)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+ g_return_val_if_fail (msg->client_rtp_ports != NULL, GST_WFD_EINVAL);
+
+ if (g_strrstr (msg->client_rtp_ports->profile, "RTP"))
+ *trans = GST_WFD_RTSP_TRANS_RTP;
+ if (g_strrstr (msg->client_rtp_ports->profile, "RDT"))
+ *trans = GST_WFD_RTSP_TRANS_RDT;
+ if (g_strrstr (msg->client_rtp_ports->profile, "AVP"))
+ *profile = GST_WFD_RTSP_PROFILE_AVP;
+ if (g_strrstr (msg->client_rtp_ports->profile, "SAVP"))
+ *profile = GST_WFD_RTSP_PROFILE_SAVP;
+ if (g_strrstr (msg->client_rtp_ports->profile, "UDP;unicast"))
+ *lowertrans = GST_WFD_RTSP_LOWER_TRANS_UDP;
+ if (g_strrstr (msg->client_rtp_ports->profile, "UDP;multicast"))
+ *lowertrans = GST_WFD_RTSP_LOWER_TRANS_UDP_MCAST;
+ if (g_strrstr (msg->client_rtp_ports->profile, "TCP;unicast"))
+ *lowertrans = GST_WFD_RTSP_LOWER_TRANS_TCP;
+ if (g_strrstr (msg->client_rtp_ports->profile, "HTTP"))
+ *lowertrans = GST_WFD_RTSP_LOWER_TRANS_HTTP;
+
+ *rtp_port0 = msg->client_rtp_ports->rtp_port0;
+ *rtp_port1 = msg->client_rtp_ports->rtp_port1;
+
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_presentation_url (GstWFDMessage * msg, gchar * wfd_url0,
+ gchar * wfd_url1)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (!msg->presentation_url)
+ msg->presentation_url = g_new0 (GstWFDPresentationUrl, 1);
+ if (wfd_url0)
+ msg->presentation_url->wfd_url0 = g_strdup (wfd_url0);
+ if (wfd_url1)
+ msg->presentation_url->wfd_url1 = g_strdup (wfd_url1);
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_presentation_url (GstWFDMessage * msg, gchar ** wfd_url0,
+ gchar ** wfd_url1)
+{
+ g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+ if (msg->presentation_url) {
+ *wfd_url0 = g_strdup (msg->presentation_url->wfd_url0);
+ *wfd_url1 = g_strdup (msg->presentation_url->wfd_url1);
+ }
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_av_format_change_timing(GstWFDMessage *msg, guint64 PTS, guint64 DTS)
+{
+ g_return_val_if_fail(msg != NULL, GST_WFD_EINVAL);
+
+ if (!msg->av_format_change_timing)
+ msg->av_format_change_timing = g_new0(GstWFDAVFormatChangeTiming, 1);
+
+ msg->av_format_change_timing->PTS = PTS;
+ msg->av_format_change_timing->DTS = DTS;
+ return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_av_format_change_timing(GstWFDMessage *msg, guint64 *PTS, guint64 *DTS)
+{
+ g_return_val_if_fail(msg != NULL, GST_WFD_EINVAL);
+ g_return_val_if_fail(PTS != NULL, GST_WFD_EINVAL);
+ g_return_val_if_fail(DTS != NULL, GST_WFD_EINVAL);
+
+ if (msg->av_format_change_timing) {
+ *PTS = msg->av_format_change_timing->PTS;
+ *DTS = msg->av_format_change_timing->DTS;
+ }
+
+ return GST_WFD_OK;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_WFD_MESSAGE_H__
+#define __GST_WFD_MESSAGE_H__
+
+#include <glib.h>
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#define GST_STRING_WFD_AUDIO_CODECS "wfd_audio_codecs"
+#define GST_STRING_WFD_VIDEO_FORMATS "wfd_video_formats"
+#define GST_STRING_WFD_3D_VIDEO_FORMATS "wfd_3d_video_formats"
+#define GST_STRING_WFD_CONTENT_PROTECTION "wfd_content_protection"
+#define GST_STRING_WFD_DISPLAY_EDID "wfd_display_edid"
+#define GST_STRING_WFD_COUPLED_SINK "wfd_coupled_sink"
+#define GST_STRING_WFD_TRIGGER_METHOD "wfd_trigger_method"
+#define GST_STRING_WFD_PRESENTATION_URL "wfd_presentation_URL"
+#define GST_STRING_WFD_CLIENT_RTP_PORTS "wfd_client_rtp_ports"
+#define GST_STRING_WFD_ROUTE "wfd_route"
+#define GST_STRING_WFD_I2C "wfd_I2C"
+#define GST_STRING_WFD_AV_FORMAT_CHANGE_TIMING "wfd_av_format_change_timing"
+#define GST_STRING_WFD_PREFERRED_DISPLAY_MODE "wfd_preferred_display_mode"
+#define GST_STRING_WFD_STANDBY_RESUME_CAPABILITY "wfd_standby_resume_capability"
+#define GST_STRING_WFD_STANDBY "wfd_standby"
+#define GST_STRING_WFD_CONNECTOR_TYPE "wfd_connector_type"
+#define GST_STRING_WFD_IDR_REQUEST "wfd_idr_request"
+
+/**
+ * GstWFDResult:
+ * @GST_WFD_OK: A successful return value
+ * @GST_WFD_EINVAL: a function was given invalid parameters
+ *
+ * Return values for the WFD functions.
+ */
+typedef enum {
+ GST_WFD_OK = 0,
+ GST_WFD_EINVAL = -1
+} GstWFDResult;
+
+
+typedef enum {
+ GST_WFD_AUDIO_UNKNOWN = 0,
+ GST_WFD_AUDIO_LPCM = (1 << 0),
+ GST_WFD_AUDIO_AAC = (1 << 1),
+ GST_WFD_AUDIO_AC3 = (1 << 2)
+} GstWFDAudioFormats;
+
+typedef enum {
+ GST_WFD_FREQ_UNKNOWN = 0,
+ GST_WFD_FREQ_44100 = (1 << 0),
+ GST_WFD_FREQ_48000 = (1 << 1)
+} GstWFDAudioFreq;
+
+typedef enum {
+ GST_WFD_CHANNEL_UNKNOWN = 0,
+ GST_WFD_CHANNEL_2 = (1 << 0),
+ GST_WFD_CHANNEL_4 = (1 << 1),
+ GST_WFD_CHANNEL_6 = (1 << 2),
+ GST_WFD_CHANNEL_8 = (1 << 3)
+} GstWFDAudioChannels;
+
+
+typedef enum {
+ GST_WFD_VIDEO_UNKNOWN = 0,
+ GST_WFD_VIDEO_H264 = (1 << 0)
+} GstWFDVideoCodecs;
+
+typedef enum {
+ GST_WFD_VIDEO_CEA_RESOLUTION = 0,
+ GST_WFD_VIDEO_VESA_RESOLUTION,
+ GST_WFD_VIDEO_HH_RESOLUTION
+} GstWFDVideoNativeResolution;
+
+typedef enum {
+ GST_WFD_CEA_UNKNOWN = 0,
+ GST_WFD_CEA_640x480P60 = (1 << 0),
+ GST_WFD_CEA_720x480P60 = (1 << 1),
+ GST_WFD_CEA_720x480I60 = (1 << 2),
+ GST_WFD_CEA_720x576P50 = (1 << 3),
+ GST_WFD_CEA_720x576I50 = (1 << 4),
+ GST_WFD_CEA_1280x720P30 = (1 << 5),
+ GST_WFD_CEA_1280x720P60 = (1 << 6),
+ GST_WFD_CEA_1920x1080P30= (1 << 7),
+ GST_WFD_CEA_1920x1080P60= (1 << 8),
+ GST_WFD_CEA_1920x1080I60= (1 << 9),
+ GST_WFD_CEA_1280x720P25 = (1 << 10),
+ GST_WFD_CEA_1280x720P50 = (1 << 11),
+ GST_WFD_CEA_1920x1080P25= (1 << 12),
+ GST_WFD_CEA_1920x1080P50= (1 << 13),
+ GST_WFD_CEA_1920x1080I50= (1 << 14),
+ GST_WFD_CEA_1280x720P24 = (1 << 15),
+ GST_WFD_CEA_1920x1080P24= (1 << 16)
+} GstWFDVideoCEAResolution;
+
+typedef enum {
+ GST_WFD_VESA_UNKNOWN = 0,
+ GST_WFD_VESA_800x600P30 = (1 << 0),
+ GST_WFD_VESA_800x600P60 = (1 << 1),
+ GST_WFD_VESA_1024x768P30 = (1 << 2),
+ GST_WFD_VESA_1024x768P60 = (1 << 3),
+ GST_WFD_VESA_1152x864P30 = (1 << 4),
+ GST_WFD_VESA_1152x864P60 = (1 << 5),
+ GST_WFD_VESA_1280x768P30 = (1 << 6),
+ GST_WFD_VESA_1280x768P60 = (1 << 7),
+ GST_WFD_VESA_1280x800P30 = (1 << 8),
+ GST_WFD_VESA_1280x800P60 = (1 << 9),
+ GST_WFD_VESA_1360x768P30 = (1 << 10),
+ GST_WFD_VESA_1360x768P60 = (1 << 11),
+ GST_WFD_VESA_1366x768P30 = (1 << 12),
+ GST_WFD_VESA_1366x768P60 = (1 << 13),
+ GST_WFD_VESA_1280x1024P30 = (1 << 14),
+ GST_WFD_VESA_1280x1024P60 = (1 << 15),
+ GST_WFD_VESA_1400x1050P30 = (1 << 16),
+ GST_WFD_VESA_1400x1050P60 = (1 << 17),
+ GST_WFD_VESA_1440x900P30 = (1 << 18),
+ GST_WFD_VESA_1440x900P60 = (1 << 19),
+ GST_WFD_VESA_1600x900P30 = (1 << 20),
+ GST_WFD_VESA_1600x900P60 = (1 << 21),
+ GST_WFD_VESA_1600x1200P30 = (1 << 22),
+ GST_WFD_VESA_1600x1200P60 = (1 << 23),
+ GST_WFD_VESA_1680x1024P30 = (1 << 24),
+ GST_WFD_VESA_1680x1024P60 = (1 << 25),
+ GST_WFD_VESA_1680x1050P30 = (1 << 26),
+ GST_WFD_VESA_1680x1050P60 = (1 << 27),
+ GST_WFD_VESA_1920x1200P30 = (1 << 28),
+ GST_WFD_VESA_1920x1200P60 = (1 << 29)
+} GstWFDVideoVESAResolution;
+
+typedef enum {
+ GST_WFD_HH_UNKNOWN = 0,
+ GST_WFD_HH_800x480P30 = (1 << 0),
+ GST_WFD_HH_800x480P60 = (1 << 1),
+ GST_WFD_HH_854x480P30 = (1 << 2),
+ GST_WFD_HH_854x480P60 = (1 << 3),
+ GST_WFD_HH_864x480P30 = (1 << 4),
+ GST_WFD_HH_864x480P60 = (1 << 5),
+ GST_WFD_HH_640x360P30 = (1 << 6),
+ GST_WFD_HH_640x360P60 = (1 << 7),
+ GST_WFD_HH_960x540P30 = (1 << 8),
+ GST_WFD_HH_960x540P60 = (1 << 9),
+ GST_WFD_HH_848x480P30 = (1 << 10),
+ GST_WFD_HH_848x480P60 = (1 << 11)
+} GstWFDVideoHHResolution;
+
+typedef enum {
+ GST_WFD_H264_UNKNOWN_PROFILE= 0,
+ GST_WFD_H264_BASE_PROFILE = (1 << 0),
+ GST_WFD_H264_HIGH_PROFILE = (1 << 1)
+} GstWFDVideoH264Profile;
+
+typedef enum {
+ GST_WFD_H264_LEVEL_UNKNOWN = 0,
+ GST_WFD_H264_LEVEL_3_1 = (1 << 0),
+ GST_WFD_H264_LEVEL_3_2 = (1 << 1),
+ GST_WFD_H264_LEVEL_4 = (1 << 2),
+ GST_WFD_H264_LEVEL_4_1 = (1 << 3),
+ GST_WFD_H264_LEVEL_4_2 = (1 << 4)
+} GstWFDVideoH264Level;
+
+typedef enum {
+ GST_WFD_HDCP_NONE = 0,
+ GST_WFD_HDCP_2_0 = (1 << 0),
+ GST_WFD_HDCP_2_1 = (1 << 1)
+} GstWFDHDCPProtection;
+
+typedef enum {
+ GST_WFD_SINK_UNKNOWN = -1,
+ GST_WFD_SINK_NOT_COUPLED = 0,
+ GST_WFD_SINK_COUPLED,
+ GST_WFD_SINK_TEARDOWN_COUPLING,
+ GST_WFD_SINK_RESERVED
+} GstWFDCoupledSinkStatus;
+
+typedef enum {
+ GST_WFD_TRIGGER_UNKNOWN = 0,
+ GST_WFD_TRIGGER_SETUP,
+ GST_WFD_TRIGGER_PAUSE,
+ GST_WFD_TRIGGER_TEARDOWN,
+ GST_WFD_TRIGGER_PLAY
+} GstWFDTrigger;
+
+typedef enum {
+ GST_WFD_RTSP_TRANS_UNKNOWN = 0,
+ GST_WFD_RTSP_TRANS_RTP = (1 << 0),
+ GST_WFD_RTSP_TRANS_RDT = (1 << 1)
+} GstWFDRTSPTransMode;
+
+typedef enum {
+ GST_WFD_RTSP_PROFILE_UNKNOWN = 0,
+ GST_WFD_RTSP_PROFILE_AVP = (1 << 0),
+ GST_WFD_RTSP_PROFILE_SAVP = (1 << 1)
+} GstWFDRTSPProfile;
+
+typedef enum {
+ GST_WFD_RTSP_LOWER_TRANS_UNKNOWN = 0,
+ GST_WFD_RTSP_LOWER_TRANS_UDP = (1 << 0),
+ GST_WFD_RTSP_LOWER_TRANS_UDP_MCAST = (1 << 1),
+ GST_WFD_RTSP_LOWER_TRANS_TCP = (1 << 2),
+ GST_WFD_RTSP_LOWER_TRANS_HTTP = (1 << 3)
+} GstWFDRTSPLowerTrans;
+
+typedef enum {
+ GST_WFD_PRIMARY_SINK = 0,
+ GST_WFD_SECONDARY_SINK
+} GstWFDSinkType;
+
+typedef enum {
+ GST_WFD_CONNECTOR_VGA = 0,
+ GST_WFD_CONNECTOR_S,
+ GST_WFD_CONNECTOR_COMPOSITE,
+ GST_WFD_CONNECTOR_COMPONENT,
+ GST_WFD_CONNECTOR_DVI,
+ GST_WFD_CONNECTOR_HDMI,
+ GST_WFD_CONNECTOR_LVDS,
+ GST_WFD_CONNECTOR_RESERVED_7,
+ GST_WFD_CONNECTOR_JAPANESE_D,
+ GST_WFD_CONNECTOR_SDI,
+ GST_WFD_CONNECTOR_DP,
+ GST_WFD_CONNECTOR_RESERVED_11,
+ GST_WFD_CONNECTOR_UDI,
+ GST_WFD_CONNECTOR_NO = 254,
+ GST_WFD_CONNECTOR_PHYSICAL = 255
+} GstWFDConnector;
+
+
+typedef struct {
+ gchar *audio_format;
+ guint32 modes;
+ guint latency;
+} GstWFDAudioCodec;
+
+typedef struct {
+ guint count;
+ GstWFDAudioCodec *list;
+} GstWFDAudioCodeclist;
+
+
+typedef struct {
+ guint CEA_Support;
+ guint VESA_Support;
+ guint HH_Support;
+ guint latency;
+ guint min_slice_size;
+ guint slice_enc_params;
+ guint frame_rate_control_support;
+} GstWFDVideoH264MiscParams;
+
+typedef struct {
+ guint profile;
+ guint level;
+ guint max_hres;
+ guint max_vres;
+ GstWFDVideoH264MiscParams misc_params;
+} GstWFDVideoH264Codec;
+
+typedef struct {
+ guint native;
+ guint preferred_display_mode_supported;
+ GstWFDVideoH264Codec H264_codec;
+} GstWFDVideoCodec;
+
+typedef struct {
+ guint count;
+ GstWFDVideoCodec *list;
+} GstWFDVideoCodeclist;
+
+typedef struct {
+ guint video_3d_capability;
+ guint latency;
+ guint min_slice_size;
+ guint slice_enc_params;
+ guint frame_rate_control_support;
+} GstWFD3DVideoH264MiscParams;
+
+typedef struct {
+ guint profile;
+ guint level;
+ GstWFD3DVideoH264MiscParams misc_params;
+ guint max_hres;
+ guint max_vres;
+} GstWFD3DVideoH264Codec;
+
+typedef struct {
+ guint native;
+ guint preferred_display_mode_supported;
+ GstWFD3DVideoH264Codec H264_codec;
+} GstWFD3dCapList;
+
+typedef struct {
+ guint count;
+ GstWFD3dCapList *list;
+} GstWFD3DFormats;
+
+typedef struct {
+ gchar *hdcpversion;
+ gchar *TCPPort;
+} GstWFDHdcp2Spec;
+
+typedef struct {
+ GstWFDHdcp2Spec *hdcp2_spec;
+} GstWFDContentProtection;
+
+typedef struct {
+ guint edid_supported;
+ guint edid_block_count;
+ gchar *edid_payload;
+} GstWFDDisplayEdid;
+
+
+typedef struct {
+ guint status;
+ gchar *sink_address;
+} GstWFDCoupled_sink_cap;
+
+typedef struct {
+ GstWFDCoupled_sink_cap *coupled_sink_cap;
+} GstWFDCoupledSink;
+
+typedef struct {
+ gchar *wfd_trigger_method;
+} GstWFDTriggerMethod;
+
+typedef struct {
+ gchar *wfd_url0;
+ gchar *wfd_url1;
+} GstWFDPresentationUrl;
+
+typedef struct {
+ gchar *profile;
+ guint32 rtp_port0;
+ guint32 rtp_port1;
+ gchar *mode;
+} GstWFDClientRtpPorts;
+
+typedef struct {
+ gchar *destination;
+} GstWFDRoute;
+
+typedef struct {
+ gboolean I2CPresent;
+ guint32 I2C_port;
+} GstWFDI2C;
+
+typedef struct {
+ guint64 PTS;
+ guint64 DTS;
+} GstWFDAVFormatChangeTiming;
+
+typedef struct {
+ gboolean displaymodesupported;
+ guint64 p_clock;
+ guint32 H;
+ guint32 HB;
+ guint32 HSPOL_HSOFF;
+ guint32 HSW;
+ guint32 V;
+ guint32 VB;
+ guint32 VSPOL_VSOFF;
+ guint32 VSW;
+ guint VBS3D;
+ guint R;
+ guint V2d_s3d_modes;
+ guint P_depth;
+ GstWFDVideoH264Codec H264_codec;
+} GstWFDPreferredDisplayMode;
+
+typedef struct {
+ gboolean standby_resume_cap;
+} GstWFDStandbyResumeCapability;
+
+typedef struct {
+ gboolean wfd_standby;
+} GstWFDStandby;
+
+typedef struct {
+ gboolean supported;
+ gint32 connector_type;
+} GstWFDConnectorType;
+
+typedef struct {
+ gboolean idr_request;
+} GstWFDIdrRequest;
+
+/**
+ * GstWFDMessage:
+ * @version: the protocol version
+ * @origin: owner/creator and session identifier
+ * @session_name: session name
+ * @information: session information
+ * @uri: URI of description
+ * @emails: array of #gchar with email addresses
+ * @phones: array of #gchar with phone numbers
+ * @connection: connection information for the session
+ * @bandwidths: array of #GstWFDBandwidth with bandwidth information
+ * @times: array of #GstWFDTime with time descriptions
+ * @zones: array of #GstWFDZone with time zone adjustments
+ * @key: encryption key
+ * @attributes: array of #GstWFDAttribute with session attributes
+ * @medias: array of #GstWFDMedia with media descriptions
+ *
+ * The contents of the WFD message.
+ */
+typedef struct {
+ GstWFDAudioCodeclist *audio_codecs;
+ GstWFDVideoCodeclist *video_formats;
+ GstWFD3DFormats *video_3d_formats;
+ GstWFDContentProtection *content_protection;
+ GstWFDDisplayEdid *display_edid;
+ GstWFDCoupledSink *coupled_sink;
+ GstWFDTriggerMethod *trigger_method;
+ GstWFDPresentationUrl *presentation_url;
+ GstWFDClientRtpPorts *client_rtp_ports;
+ GstWFDRoute *route;
+ GstWFDI2C *I2C;
+ GstWFDAVFormatChangeTiming *av_format_change_timing;
+ GstWFDPreferredDisplayMode *preferred_display_mode;
+ GstWFDStandbyResumeCapability *standby_resume_capability;
+ GstWFDStandby *standby;
+ GstWFDConnectorType *connector_type;
+ GstWFDIdrRequest *idr_request;
+} GstWFDMessage;
+
+GType gst_wfd_message_get_type (void);
+
+#define GST_TYPE_WFD_MESSAGE (gst_wfd_message_get_type())
+#define GST_WFD_MESSAGE_CAST(object) ((GstWFDMessage *)(object))
+#define GST_WFD_MESSAGE(object) (GST_WFD_MESSAGE_CAST(object))
+
+/* Session descriptions */
+GstWFDResult gst_wfd_message_new (GstWFDMessage **msg);
+GstWFDResult gst_wfd_message_init (GstWFDMessage *msg);
+GstWFDResult gst_wfd_message_uninit (GstWFDMessage *msg);
+GstWFDResult gst_wfd_message_free (GstWFDMessage *msg);
+GstWFDResult gst_wfd_message_copy (const GstWFDMessage *msg, GstWFDMessage **copy);
+
+GstWFDResult gst_wfd_message_parse_buffer (const guint8 *data, guint size, GstWFDMessage *msg);
+gchar* gst_wfd_message_as_text (const GstWFDMessage *msg);
+gchar* gst_wfd_message_param_names_as_text (const GstWFDMessage *msg);
+GstWFDResult gst_wfd_message_dump (const GstWFDMessage *msg);
+
+
+GstWFDResult gst_wfd_message_set_supported_audio_format(GstWFDMessage *msg,
+ GstWFDAudioFormats a_codec,
+ guint a_freq, guint a_channels,
+ guint a_bitwidth, guint32 a_latency);
+
+GstWFDResult gst_wfd_message_set_prefered_audio_format(GstWFDMessage *msg,
+ GstWFDAudioFormats a_codec,
+ GstWFDAudioFreq a_freq,
+ GstWFDAudioChannels a_channels,
+ guint a_bitwidth, guint32 a_latency);
+
+GstWFDResult gst_wfd_message_get_supported_audio_format (GstWFDMessage *msg,
+ guint *a_codec,
+ guint *a_freq,
+ guint *a_channels,
+ guint *a_bitwidth,
+ guint32 *a_latency);
+
+GstWFDResult gst_wfd_message_get_prefered_audio_format (GstWFDMessage *msg,
+ GstWFDAudioFormats *a_codec,
+ GstWFDAudioFreq *a_freq,
+ GstWFDAudioChannels *a_channels,
+ guint *a_bitwidth, guint32 *a_latency);
+
+GstWFDResult gst_wfd_message_set_supported_video_format (GstWFDMessage *msg,
+ GstWFDVideoCodecs v_codec,
+ GstWFDVideoNativeResolution v_native,
+ guint64 v_native_resolution,
+ guint64 v_cea_resolution,
+ guint64 v_vesa_resolution,
+ guint64 v_hh_resolution,
+ guint v_profile,
+ guint v_level,
+ guint32 v_latency,
+ guint32 v_max_height,
+ guint32 v_max_width,
+ guint32 min_slice_size,
+ guint32 slice_enc_params,
+ guint frame_rate_control);
+
+GstWFDResult gst_wfd_message_set_prefered_video_format(GstWFDMessage *msg,
+ GstWFDVideoCodecs v_codec,
+ GstWFDVideoNativeResolution v_native,
+ guint64 v_native_resolution,
+ GstWFDVideoCEAResolution v_cea_resolution,
+ GstWFDVideoVESAResolution v_vesa_resolution,
+ GstWFDVideoHHResolution v_hh_resolution,
+ GstWFDVideoH264Profile v_profile,
+ GstWFDVideoH264Level v_level,
+ guint32 v_latency,
+ guint32 v_max_height,
+ guint32 v_max_width,
+ guint32 min_slice_size,
+ guint32 slice_enc_params,
+ guint frame_rate_control);
+
+GstWFDResult gst_wfd_message_get_supported_video_format(GstWFDMessage *msg,
+ GstWFDVideoCodecs *v_codec,
+ GstWFDVideoNativeResolution *v_native,
+ guint64 *v_native_resolution,
+ guint64 *v_cea_resolution,
+ guint64 *v_vesa_resolution,
+ guint64 *v_hh_resolution,
+ guint *v_profile,
+ guint *v_level,
+ guint32 *v_latency,
+ guint32 *v_max_height,
+ guint32 *v_max_width,
+ guint32 *min_slice_size,
+ guint32 *slice_enc_params,
+ guint *frame_rate_control);
+
+GstWFDResult gst_wfd_message_get_prefered_video_format(GstWFDMessage *msg,
+ GstWFDVideoCodecs *v_codec,
+ GstWFDVideoNativeResolution *v_native,
+ guint64 *v_native_resolution,
+ GstWFDVideoCEAResolution *v_cea_resolution,
+ GstWFDVideoVESAResolution *v_vesa_resolution,
+ GstWFDVideoHHResolution *v_hh_resolution,
+ GstWFDVideoH264Profile *v_profile,
+ GstWFDVideoH264Level *v_level,
+ guint32 *v_latency,
+ guint32 *v_max_height,
+ guint32 *v_max_width,
+ guint32 *min_slice_size,
+ guint32 *slice_enc_params,
+ guint *frame_rate_control);
+
+GstWFDResult gst_wfd_message_set_display_edid (GstWFDMessage *msg,
+ gboolean edid_supported,
+ guint32 edid_blockcount,
+ gchar *edid_playload);
+
+GstWFDResult gst_wfd_message_get_display_edid (GstWFDMessage *msg,
+ gboolean *edid_supported,
+ guint32 *edid_blockcount,
+ gchar **edid_playload);
+
+GstWFDResult gst_wfd_message_set_contentprotection_type (GstWFDMessage *msg,
+ GstWFDHDCPProtection hdcpversion,
+ guint32 TCPPort);
+
+GstWFDResult gst_wfd_message_get_contentprotection_type (GstWFDMessage *msg,
+ GstWFDHDCPProtection *hdcpversion,
+ guint32 *TCPPort);
+
+GstWFDResult gst_wfd_messge_set_prefered_rtp_ports (GstWFDMessage *msg,
+ GstWFDRTSPTransMode trans,
+ GstWFDRTSPProfile profile,
+ GstWFDRTSPLowerTrans lowertrans,
+ guint32 rtp_port0,
+ guint32 rtp_port1);
+
+GstWFDResult gst_wfd_message_get_prefered_rtp_ports (GstWFDMessage *msg,
+ GstWFDRTSPTransMode *trans,
+ GstWFDRTSPProfile *profile,
+ GstWFDRTSPLowerTrans *lowertrans,
+ guint32 *rtp_port0,
+ guint32 *rtp_port1);
+
+GstWFDResult gst_wfd_message_set_presentation_url(GstWFDMessage *msg,
+ gchar *wfd_url0, gchar *wfd_url1);
+
+GstWFDResult gst_wfd_message_get_presentation_url(GstWFDMessage *msg, gchar **wfd_url0,
+ gchar **wfd_url1);
+
+GstWFDResult gst_wfd_message_set_av_format_change_timing(GstWFDMessage *msg,
+ guint64 PTS,
+ guint64 DTS);
+
+GstWFDResult gst_wfd_message_get_av_format_change_timing(GstWFDMessage *msg,
+ guint64 *PTS,
+ guint64 *DTS);
+G_END_DECLS
+
+#endif /* __GST_WFD_MESSAGE_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-address-pool
+ * @short_description: A pool of network addresses
+ * @see_also: #GstRTSPStream, #GstRTSPStreamTransport
+ *
+ * The #GstRTSPAddressPool is an object that maintains a collection of network
+ * addresses. It is used to allocate server ports and server multicast addresses
+ * but also to reserve client provided destination addresses.
+ *
+ * A range of addresses can be added with gst_rtsp_address_pool_add_range().
+ * Both multicast and unicast addresses can be added.
+ *
+ * With gst_rtsp_address_pool_acquire_address() an unused address and port range
+ * can be acquired from the pool. With gst_rtsp_address_pool_reserve_address() a
+ * specific address can be retrieved. Both methods return a boxed
+ * #GstRTSPAddress that should be freed with gst_rtsp_address_free() after
+ * usage, which brings the address back into the pool.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+
+#include <string.h>
+#include <gio/gio.h>
+
+#include "rtsp-address-pool.h"
+
+/**
+ * gst_rtsp_address_copy:
+ * @addr: a #GstRTSPAddress
+ *
+ * Make a copy of @addr.
+ *
+ * Returns: a copy of @addr.
+ */
+GstRTSPAddress *
+gst_rtsp_address_copy (GstRTSPAddress * addr)
+{
+ GstRTSPAddress *copy;
+
+ g_return_val_if_fail (addr != NULL, NULL);
+
+ copy = g_slice_dup (GstRTSPAddress, addr);
+ /* only release to the pool when the original is freed. It's a bit
+ * weird but this will do for now as it avoid us to use refcounting. */
+ copy->pool = NULL;
+ copy->address = g_strdup (copy->address);
+
+ return copy;
+}
+
+static void gst_rtsp_address_pool_release_address (GstRTSPAddressPool * pool,
+ GstRTSPAddress * addr);
+
+/**
+ * gst_rtsp_address_free:
+ * @addr: a #GstRTSPAddress
+ *
+ * Free @addr and releasing it back into the pool when owned by a
+ * pool.
+ */
+void
+gst_rtsp_address_free (GstRTSPAddress * addr)
+{
+ g_return_if_fail (addr != NULL);
+
+ if (addr->pool) {
+ /* unrefs the pool and sets it to NULL */
+ gst_rtsp_address_pool_release_address (addr->pool, addr);
+ }
+ g_free (addr->address);
+ g_slice_free (GstRTSPAddress, addr);
+}
+
+G_DEFINE_BOXED_TYPE (GstRTSPAddress, gst_rtsp_address,
+ (GBoxedCopyFunc) gst_rtsp_address_copy,
+ (GBoxedFreeFunc) gst_rtsp_address_free);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_address_pool_debug);
+#define GST_CAT_DEFAULT rtsp_address_pool_debug
+
+#define GST_RTSP_ADDRESS_POOL_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPoolPrivate))
+
+struct _GstRTSPAddressPoolPrivate
+{
+ GMutex lock; /* protects everything in this struct */
+ GList *addresses;
+ GList *allocated;
+
+ gboolean has_unicast_addresses;
+};
+
+#define ADDR_IS_IPV4(a) ((a)->size == 4)
+#define ADDR_IS_IPV6(a) ((a)->size == 16)
+#define ADDR_IS_EVEN_PORT(a) (((a)->port & 1) == 0)
+
+typedef struct
+{
+ guint8 bytes[16];
+ gsize size;
+ guint16 port;
+} Addr;
+
+typedef struct
+{
+ Addr min;
+ Addr max;
+ guint8 ttl;
+} AddrRange;
+
+#define RANGE_IS_SINGLE(r) (memcmp ((r)->min.bytes, (r)->max.bytes, (r)->min.size) == 0)
+
+#define gst_rtsp_address_pool_parent_class parent_class
+G_DEFINE_TYPE (GstRTSPAddressPool, gst_rtsp_address_pool, G_TYPE_OBJECT);
+
+static void gst_rtsp_address_pool_finalize (GObject * obj);
+
+static void
+gst_rtsp_address_pool_class_init (GstRTSPAddressPoolClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->finalize = gst_rtsp_address_pool_finalize;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPAddressPoolPrivate));
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_address_pool_debug, "rtspaddresspool", 0,
+ "GstRTSPAddressPool");
+}
+
+static void
+gst_rtsp_address_pool_init (GstRTSPAddressPool * pool)
+{
+ pool->priv = GST_RTSP_ADDRESS_POOL_GET_PRIVATE (pool);
+
+ g_mutex_init (&pool->priv->lock);
+}
+
+static void
+free_range (AddrRange * range)
+{
+ g_slice_free (AddrRange, range);
+}
+
+static void
+gst_rtsp_address_pool_finalize (GObject * obj)
+{
+ GstRTSPAddressPool *pool;
+
+ pool = GST_RTSP_ADDRESS_POOL (obj);
+
+ g_list_free_full (pool->priv->addresses, (GDestroyNotify) free_range);
+ g_list_free_full (pool->priv->allocated, (GDestroyNotify) free_range);
+ g_mutex_clear (&pool->priv->lock);
+
+ G_OBJECT_CLASS (parent_class)->finalize (obj);
+}
+
+/**
+ * gst_rtsp_address_pool_new:
+ *
+ * Make a new #GstRTSPAddressPool.
+ *
+ * Returns: (transfer full): a new #GstRTSPAddressPool
+ */
+GstRTSPAddressPool *
+gst_rtsp_address_pool_new (void)
+{
+ GstRTSPAddressPool *pool;
+
+ pool = g_object_new (GST_TYPE_RTSP_ADDRESS_POOL, NULL);
+
+ return pool;
+}
+
+/**
+ * gst_rtsp_address_pool_clear:
+ * @pool: a #GstRTSPAddressPool
+ *
+ * Clear all addresses in @pool. There should be no outstanding
+ * allocations.
+ */
+void
+gst_rtsp_address_pool_clear (GstRTSPAddressPool * pool)
+{
+ GstRTSPAddressPoolPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool));
+ g_return_if_fail (pool->priv->allocated == NULL);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_list_free_full (priv->addresses, (GDestroyNotify) free_range);
+ priv->addresses = NULL;
+ g_mutex_unlock (&priv->lock);
+}
+
+static gboolean
+fill_address (const gchar * address, guint16 port, Addr * addr,
+ gboolean is_multicast)
+{
+ GInetAddress *inet;
+
+ inet = g_inet_address_new_from_string (address);
+ if (inet == NULL)
+ return FALSE;
+
+ if (is_multicast != g_inet_address_get_is_multicast (inet)) {
+ g_object_unref (inet);
+ return FALSE;
+ }
+
+ addr->size = g_inet_address_get_native_size (inet);
+ memcpy (addr->bytes, g_inet_address_to_bytes (inet), addr->size);
+ g_object_unref (inet);
+ addr->port = port;
+
+ return TRUE;
+}
+
+static gchar *
+get_address_string (Addr * addr)
+{
+ gchar *res;
+ GInetAddress *inet;
+
+ inet = g_inet_address_new_from_bytes (addr->bytes,
+ addr->size == 4 ? G_SOCKET_FAMILY_IPV4 : G_SOCKET_FAMILY_IPV6);
+ res = g_inet_address_to_string (inet);
+ g_object_unref (inet);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_address_pool_add_range:
+ * @pool: a #GstRTSPAddressPool
+ * @min_address: a minimum address to add
+ * @max_address: a maximum address to add
+ * @min_port: the minimum port
+ * @max_port: the maximum port
+ * @ttl: a TTL or 0 for unicast addresses
+ *
+ * Adds the addresses from @min_addess to @max_address (inclusive)
+ * to @pool. The valid port range for the addresses will be from @min_port to
+ * @max_port inclusive.
+ *
+ * When @ttl is 0, @min_address and @max_address should be unicast addresses.
+ * @min_address and @max_address can be set to
+ * #GST_RTSP_ADDRESS_POOL_ANY_IPV4 or #GST_RTSP_ADDRESS_POOL_ANY_IPV6 to bind
+ * to all available IPv4 or IPv6 addresses.
+ *
+ * When @ttl > 0, @min_address and @max_address should be multicast addresses.
+ *
+ * Returns: %TRUE if the addresses could be added.
+ */
+gboolean
+gst_rtsp_address_pool_add_range (GstRTSPAddressPool * pool,
+ const gchar * min_address, const gchar * max_address,
+ guint16 min_port, guint16 max_port, guint8 ttl)
+{
+ AddrRange *range;
+ GstRTSPAddressPoolPrivate *priv;
+ gboolean is_multicast;
+
+ g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool), FALSE);
+ g_return_val_if_fail (min_port <= max_port, FALSE);
+
+ priv = pool->priv;
+
+ is_multicast = ttl != 0;
+
+ range = g_slice_new0 (AddrRange);
+
+ if (!fill_address (min_address, min_port, &range->min, is_multicast))
+ goto invalid;
+ if (!fill_address (max_address, max_port, &range->max, is_multicast))
+ goto invalid;
+
+ if (range->min.size != range->max.size)
+ goto invalid;
+ if (memcmp (range->min.bytes, range->max.bytes, range->min.size) > 0)
+ goto invalid;
+
+ range->ttl = ttl;
+
+ GST_DEBUG_OBJECT (pool, "adding %s-%s:%u-%u ttl %u", min_address, max_address,
+ min_port, max_port, ttl);
+
+ g_mutex_lock (&priv->lock);
+ priv->addresses = g_list_prepend (priv->addresses, range);
+
+ if (!is_multicast)
+ priv->has_unicast_addresses = TRUE;
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+ /* ERRORS */
+invalid:
+ {
+ GST_ERROR_OBJECT (pool, "invalid address range %s-%s", min_address,
+ max_address);
+ g_slice_free (AddrRange, range);
+ return FALSE;
+ }
+}
+
+static void
+inc_address (Addr * addr, guint count)
+{
+ gint i;
+ guint carry;
+
+ carry = count;
+ for (i = addr->size - 1; i >= 0 && carry > 0; i--) {
+ carry += addr->bytes[i];
+ addr->bytes[i] = carry & 0xff;
+ carry >>= 8;
+ }
+}
+
+/* tells us the number of addresses between min_addr and max_addr */
+static guint
+diff_address (Addr * max_addr, Addr * min_addr)
+{
+ gint i;
+ guint result = 0;
+
+ g_return_val_if_fail (min_addr->size == max_addr->size, 0);
+
+ for (i = 0; i < min_addr->size; i++) {
+ g_return_val_if_fail (result < (1 << 24), result);
+
+ result <<= 8;
+ result += max_addr->bytes[i] - min_addr->bytes[i];
+ }
+
+ return result;
+}
+
+static AddrRange *
+split_range (GstRTSPAddressPool * pool, AddrRange * range, guint skip_addr,
+ guint skip_port, gint n_ports)
+{
+ GstRTSPAddressPoolPrivate *priv = pool->priv;
+ AddrRange *temp;
+
+ if (skip_addr) {
+ temp = g_slice_dup (AddrRange, range);
+ memcpy (temp->max.bytes, temp->min.bytes, temp->min.size);
+ inc_address (&temp->max, skip_addr - 1);
+ priv->addresses = g_list_prepend (priv->addresses, temp);
+
+ inc_address (&range->min, skip_addr);
+ }
+
+ if (!RANGE_IS_SINGLE (range)) {
+ /* min and max are not the same, we have more than one address. */
+ temp = g_slice_dup (AddrRange, range);
+ /* increment the range min address */
+ inc_address (&temp->min, 1);
+ /* and store back in pool */
+ priv->addresses = g_list_prepend (priv->addresses, temp);
+
+ /* adjust range with only the first address */
+ memcpy (range->max.bytes, range->min.bytes, range->min.size);
+ }
+
+ /* range now contains only one single address */
+ if (skip_port > 0) {
+ /* make a range with the skipped ports */
+ temp = g_slice_dup (AddrRange, range);
+ temp->max.port = temp->min.port + skip_port - 1;
+ /* and store back in pool */
+ priv->addresses = g_list_prepend (priv->addresses, temp);
+
+ /* increment range port */
+ range->min.port += skip_port;
+ }
+ /* range now contains single address with desired port number */
+ if (range->max.port - range->min.port + 1 > n_ports) {
+ /* make a range with the remaining ports */
+ temp = g_slice_dup (AddrRange, range);
+ temp->min.port += n_ports;
+ /* and store back in pool */
+ priv->addresses = g_list_prepend (priv->addresses, temp);
+
+ /* and truncate port */
+ range->max.port = range->min.port + n_ports - 1;
+ }
+ return range;
+}
+
+/**
+ * gst_rtsp_address_pool_acquire_address:
+ * @pool: a #GstRTSPAddressPool
+ * @flags: flags
+ * @n_ports: the amount of ports
+ *
+ * Take an address and ports from @pool. @flags can be used to control the
+ * allocation. @n_ports consecutive ports will be allocated of which the first
+ * one can be found in @port.
+ *
+ * Returns: (nullable): a #GstRTSPAddress that should be freed with
+ * gst_rtsp_address_free after use or %NULL when no address could be
+ * acquired.
+ */
+GstRTSPAddress *
+gst_rtsp_address_pool_acquire_address (GstRTSPAddressPool * pool,
+ GstRTSPAddressFlags flags, gint n_ports)
+{
+ GstRTSPAddressPoolPrivate *priv;
+ GList *walk, *next;
+ AddrRange *result;
+ GstRTSPAddress *addr;
+
+ g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool), NULL);
+ g_return_val_if_fail (n_ports > 0, NULL);
+
+ priv = pool->priv;
+ result = NULL;
+ addr = NULL;
+
+ g_mutex_lock (&priv->lock);
+ /* go over available ranges */
+ for (walk = priv->addresses; walk; walk = next) {
+ AddrRange *range;
+ gint ports, skip;
+
+ range = walk->data;
+ next = walk->next;
+
+ /* check address type when given */
+ if (flags & GST_RTSP_ADDRESS_FLAG_IPV4 && !ADDR_IS_IPV4 (&range->min))
+ continue;
+ if (flags & GST_RTSP_ADDRESS_FLAG_IPV6 && !ADDR_IS_IPV6 (&range->min))
+ continue;
+ if (flags & GST_RTSP_ADDRESS_FLAG_MULTICAST && range->ttl == 0)
+ continue;
+ if (flags & GST_RTSP_ADDRESS_FLAG_UNICAST && range->ttl != 0)
+ continue;
+
+ /* check for enough ports */
+ ports = range->max.port - range->min.port + 1;
+ if (flags & GST_RTSP_ADDRESS_FLAG_EVEN_PORT
+ && !ADDR_IS_EVEN_PORT (&range->min))
+ skip = 1;
+ else
+ skip = 0;
+ if (ports - skip < n_ports)
+ continue;
+
+ /* we found a range, remove from the list */
+ priv->addresses = g_list_delete_link (priv->addresses, walk);
+ /* now split and exit our loop */
+ result = split_range (pool, range, 0, skip, n_ports);
+ priv->allocated = g_list_prepend (priv->allocated, result);
+ break;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (result) {
+ addr = g_slice_new0 (GstRTSPAddress);
+ addr->pool = g_object_ref (pool);
+ addr->address = get_address_string (&result->min);
+ addr->n_ports = n_ports;
+ addr->port = result->min.port;
+ addr->ttl = result->ttl;
+ addr->priv = result;
+
+ GST_DEBUG_OBJECT (pool, "got address %s:%u ttl %u", addr->address,
+ addr->port, addr->ttl);
+ }
+
+ return addr;
+}
+
+/**
+ * gst_rtsp_address_pool_release_address:
+ * @pool: a #GstRTSPAddressPool
+ * @id: an address id
+ *
+ * Release a previously acquired address (with
+ * gst_rtsp_address_pool_acquire_address()) back into @pool.
+ */
+static void
+gst_rtsp_address_pool_release_address (GstRTSPAddressPool * pool,
+ GstRTSPAddress * addr)
+{
+ GstRTSPAddressPoolPrivate *priv;
+ GList *find;
+ AddrRange *range;
+
+ g_return_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool));
+ g_return_if_fail (addr != NULL);
+ g_return_if_fail (addr->pool == pool);
+
+ priv = pool->priv;
+ range = addr->priv;
+
+ /* we don't want to free twice */
+ addr->priv = NULL;
+ addr->pool = NULL;
+
+ g_mutex_lock (&priv->lock);
+ find = g_list_find (priv->allocated, range);
+ if (find == NULL)
+ goto not_found;
+
+ priv->allocated = g_list_delete_link (priv->allocated, find);
+
+ /* FIXME, merge and do something clever */
+ priv->addresses = g_list_prepend (priv->addresses, range);
+ g_mutex_unlock (&priv->lock);
+
+ g_object_unref (pool);
+
+ return;
+
+ /* ERRORS */
+not_found:
+ {
+ g_warning ("Released unknown address %p", addr);
+ g_mutex_unlock (&priv->lock);
+ return;
+ }
+}
+
+static void
+dump_range (AddrRange * range, GstRTSPAddressPool * pool)
+{
+ gchar *addr1, *addr2;
+
+ addr1 = get_address_string (&range->min);
+ addr2 = get_address_string (&range->max);
+ g_print (" address %s-%s, port %u-%u, ttl %u\n", addr1, addr2,
+ range->min.port, range->max.port, range->ttl);
+ g_free (addr1);
+ g_free (addr2);
+}
+
+/**
+ * gst_rtsp_address_pool_dump:
+ * @pool: a #GstRTSPAddressPool
+ *
+ * Dump the free and allocated addresses to stdout.
+ */
+void
+gst_rtsp_address_pool_dump (GstRTSPAddressPool * pool)
+{
+ GstRTSPAddressPoolPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool));
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_print ("free:\n");
+ g_list_foreach (priv->addresses, (GFunc) dump_range, pool);
+ g_print ("allocated:\n");
+ g_list_foreach (priv->allocated, (GFunc) dump_range, pool);
+ g_mutex_unlock (&priv->lock);
+}
+
+static GList *
+find_address_in_ranges (GList * addresses, Addr * addr, guint port,
+ guint n_ports, guint ttl)
+{
+ GList *walk, *next;
+
+ /* go over available ranges */
+ for (walk = addresses; walk; walk = next) {
+ AddrRange *range;
+
+ range = walk->data;
+ next = walk->next;
+
+ /* Not the right type of address */
+ if (range->min.size != addr->size)
+ continue;
+
+ /* Check that the address is in the interval */
+ if (memcmp (range->min.bytes, addr->bytes, addr->size) > 0 ||
+ memcmp (range->max.bytes, addr->bytes, addr->size) < 0)
+ continue;
+
+ /* Make sure the requested ports are inside the range */
+ if (port < range->min.port || port + n_ports - 1 > range->max.port)
+ continue;
+
+ if (ttl != range->ttl)
+ continue;
+
+ break;
+ }
+
+ return walk;
+}
+
+/**
+ * gst_rtsp_address_pool_reserve_address:
+ * @pool: a #GstRTSPAddressPool
+ * @ip_address: The IP address to reserve
+ * @port: The first port to reserve
+ * @n_ports: The number of ports
+ * @ttl: The requested ttl
+ * @address: (out): storage for a #GstRTSPAddress
+ *
+ * Take a specific address and ports from @pool. @n_ports consecutive
+ * ports will be allocated of which the first one can be found in
+ * @port.
+ *
+ * If @ttl is 0, @address should be a unicast address. If @ttl > 0, @address
+ * should be a valid multicast address.
+ *
+ * Returns: #GST_RTSP_ADDRESS_POOL_OK if an address was reserved. The address
+ * is returned in @address and should be freed with gst_rtsp_address_free
+ * after use.
+ */
+GstRTSPAddressPoolResult
+gst_rtsp_address_pool_reserve_address (GstRTSPAddressPool * pool,
+ const gchar * ip_address, guint port, guint n_ports, guint ttl,
+ GstRTSPAddress ** address)
+{
+ GstRTSPAddressPoolPrivate *priv;
+ Addr input_addr;
+ GList *list;
+ AddrRange *addr_range;
+ GstRTSPAddress *addr;
+ gboolean is_multicast;
+ GstRTSPAddressPoolResult result;
+
+ g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool),
+ GST_RTSP_ADDRESS_POOL_EINVAL);
+ g_return_val_if_fail (ip_address != NULL, GST_RTSP_ADDRESS_POOL_EINVAL);
+ g_return_val_if_fail (port > 0, GST_RTSP_ADDRESS_POOL_EINVAL);
+ g_return_val_if_fail (n_ports > 0, GST_RTSP_ADDRESS_POOL_EINVAL);
+ g_return_val_if_fail (address != NULL, GST_RTSP_ADDRESS_POOL_EINVAL);
+
+ priv = pool->priv;
+ addr_range = NULL;
+ addr = NULL;
+ is_multicast = ttl != 0;
+
+ if (!fill_address (ip_address, port, &input_addr, is_multicast))
+ goto invalid;
+
+ g_mutex_lock (&priv->lock);
+ list = find_address_in_ranges (priv->addresses, &input_addr, port, n_ports,
+ ttl);
+ if (list != NULL) {
+ AddrRange *range = list->data;
+ guint skip_port, skip_addr;
+
+ skip_addr = diff_address (&input_addr, &range->min);
+ skip_port = port - range->min.port;
+
+ GST_DEBUG_OBJECT (pool, "diff 0x%08x/%u", skip_addr, skip_port);
+
+ /* we found a range, remove from the list */
+ priv->addresses = g_list_delete_link (priv->addresses, list);
+ /* now split and exit our loop */
+ addr_range = split_range (pool, range, skip_addr, skip_port, n_ports);
+ priv->allocated = g_list_prepend (priv->allocated, addr_range);
+ }
+
+ if (addr_range) {
+ addr = g_slice_new0 (GstRTSPAddress);
+ addr->pool = g_object_ref (pool);
+ addr->address = get_address_string (&addr_range->min);
+ addr->n_ports = n_ports;
+ addr->port = addr_range->min.port;
+ addr->ttl = addr_range->ttl;
+ addr->priv = addr_range;
+
+ result = GST_RTSP_ADDRESS_POOL_OK;
+ GST_DEBUG_OBJECT (pool, "reserved address %s:%u ttl %u", addr->address,
+ addr->port, addr->ttl);
+ } else {
+ /* We failed to reserve the address. Check if it was because the address
+ * was already in use or if it wasn't in the pool to begin with */
+ list = find_address_in_ranges (priv->allocated, &input_addr, port, n_ports,
+ ttl);
+ if (list != NULL) {
+ result = GST_RTSP_ADDRESS_POOL_ERESERVED;
+ } else {
+ result = GST_RTSP_ADDRESS_POOL_ERANGE;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ *address = addr;
+ return result;
+
+ /* ERRORS */
+invalid:
+ {
+ GST_ERROR_OBJECT (pool, "invalid address %s:%u/%u/%u", ip_address,
+ port, n_ports, ttl);
+ *address = NULL;
+ return GST_RTSP_ADDRESS_POOL_EINVAL;
+ }
+}
+
+/**
+ * gst_rtsp_address_pool_has_unicast_addresses:
+ * @pool: a #GstRTSPAddressPool
+ *
+ * Used to know if the pool includes any unicast addresses.
+ *
+ * Returns: %TRUE if the pool includes any unicast addresses, %FALSE otherwise
+ */
+
+gboolean
+gst_rtsp_address_pool_has_unicast_addresses (GstRTSPAddressPool * pool)
+{
+ GstRTSPAddressPoolPrivate *priv;
+ gboolean has_unicast_addresses;
+
+ g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool), FALSE);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ has_unicast_addresses = priv->has_unicast_addresses;
+ g_mutex_unlock (&priv->lock);
+
+ return has_unicast_addresses;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_ADDRESS_POOL_H__
+#define __GST_RTSP_ADDRESS_POOL_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_ADDRESS_POOL (gst_rtsp_address_pool_get_type ())
+#define GST_IS_RTSP_ADDRESS_POOL(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_ADDRESS_POOL))
+#define GST_IS_RTSP_ADDRESS_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_ADDRESS_POOL))
+#define GST_RTSP_ADDRESS_POOL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPoolClass))
+#define GST_RTSP_ADDRESS_POOL(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPool))
+#define GST_RTSP_ADDRESS_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPoolClass))
+#define GST_RTSP_ADDRESS_POOL_CAST(obj) ((GstRTSPAddressPool*)(obj))
+#define GST_RTSP_ADDRESS_POOL_CLASS_CAST(klass) ((GstRTSPAddressPoolClass*)(klass))
+
+/**
+ * GstRTSPAddressPoolResult:
+ * @GST_RTSP_ADDRESS_POOL_OK: no error
+ * @GST_RTSP_ADDRESS_POOL_EINVAL:invalid arguments were provided to a function
+ * @GST_RTSP_ADDRESS_POOL_ERESERVED: the addres has already been reserved
+ * @GST_RTSP_ADDRESS_POOL_ERANGE: the address is not in the pool
+ * @GST_RTSP_ADDRESS_POOL_ELAST: last error
+ *
+ * Result codes from RTSP address pool functions.
+ */
+typedef enum {
+ GST_RTSP_ADDRESS_POOL_OK = 0,
+ /* errors */
+ GST_RTSP_ADDRESS_POOL_EINVAL = -1,
+ GST_RTSP_ADDRESS_POOL_ERESERVED = -2,
+ GST_RTSP_ADDRESS_POOL_ERANGE = -3,
+
+ GST_RTSP_ADDRESS_POOL_ELAST = -4,
+} GstRTSPAddressPoolResult;
+
+
+typedef struct _GstRTSPAddress GstRTSPAddress;
+
+typedef struct _GstRTSPAddressPool GstRTSPAddressPool;
+typedef struct _GstRTSPAddressPoolClass GstRTSPAddressPoolClass;
+typedef struct _GstRTSPAddressPoolPrivate GstRTSPAddressPoolPrivate;
+
+/**
+ * GstRTSPAddress:
+ * @pool: the #GstRTSPAddressPool owner of this address
+ * @address: the address
+ * @port: the port number
+ * @n_ports: number of ports
+ * @ttl: TTL or 0 for unicast addresses
+ *
+ * An address
+ */
+struct _GstRTSPAddress {
+ GstRTSPAddressPool *pool;
+
+ gchar *address;
+ guint16 port;
+ gint n_ports;
+ guint8 ttl;
+
+ /*<private >*/
+ gpointer priv;
+};
+
+GType gst_rtsp_address_get_type (void);
+
+GstRTSPAddress * gst_rtsp_address_copy (GstRTSPAddress *addr);
+void gst_rtsp_address_free (GstRTSPAddress *addr);
+
+/**
+ * GstRTSPAddressFlags:
+ * @GST_RTSP_ADDRESS_FLAG_NONE: no flags
+ * @GST_RTSP_ADDRESS_FLAG_IPV4: an IPv4 address
+ * @GST_RTSP_ADDRESS_FLAG_IPV6: and IPv6 address
+ * @GST_RTSP_ADDRESS_FLAG_EVEN_PORT: address with an even port
+ * @GST_RTSP_ADDRESS_FLAG_MULTICAST: a multicast address
+ * @GST_RTSP_ADDRESS_FLAG_UNICAST: a unicast address
+ *
+ * Flags used to control allocation of addresses
+ */
+typedef enum {
+ GST_RTSP_ADDRESS_FLAG_NONE = 0,
+ GST_RTSP_ADDRESS_FLAG_IPV4 = (1 << 0),
+ GST_RTSP_ADDRESS_FLAG_IPV6 = (1 << 1),
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT = (1 << 2),
+ GST_RTSP_ADDRESS_FLAG_MULTICAST = (1 << 3),
+ GST_RTSP_ADDRESS_FLAG_UNICAST = (1 << 4),
+} GstRTSPAddressFlags;
+
+/**
+ * GST_RTSP_ADDRESS_POOL_ANY_IPV4:
+ *
+ * Used with gst_rtsp_address_pool_add_range() to bind to all
+ * IPv4 addresses
+ */
+#define GST_RTSP_ADDRESS_POOL_ANY_IPV4 "0.0.0.0"
+
+/**
+ * GST_RTSP_ADDRESS_POOL_ANY_IPV6:
+ *
+ * Used with gst_rtsp_address_pool_add_range() to bind to all
+ * IPv6 addresses
+ */
+#define GST_RTSP_ADDRESS_POOL_ANY_IPV6 "::"
+
+/**
+ * GstRTSPAddressPool:
+ * @parent: the parent GObject
+ *
+ * An address pool, all member are private
+ */
+struct _GstRTSPAddressPool {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPAddressPoolPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPAddressPoolClass:
+ *
+ * Opaque Address pool class.
+ */
+struct _GstRTSPAddressPoolClass {
+ GObjectClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_address_pool_get_type (void);
+
+/* create a new address pool */
+GstRTSPAddressPool * gst_rtsp_address_pool_new (void);
+
+void gst_rtsp_address_pool_clear (GstRTSPAddressPool * pool);
+void gst_rtsp_address_pool_dump (GstRTSPAddressPool * pool);
+
+gboolean gst_rtsp_address_pool_add_range (GstRTSPAddressPool * pool,
+ const gchar *min_address,
+ const gchar *max_address,
+ guint16 min_port,
+ guint16 max_port,
+ guint8 ttl);
+
+GstRTSPAddress * gst_rtsp_address_pool_acquire_address (GstRTSPAddressPool * pool,
+ GstRTSPAddressFlags flags,
+ gint n_ports);
+
+GstRTSPAddressPoolResult gst_rtsp_address_pool_reserve_address (GstRTSPAddressPool * pool,
+ const gchar *ip_address,
+ guint port,
+ guint n_ports,
+ guint ttl,
+ GstRTSPAddress ** address);
+
+gboolean gst_rtsp_address_pool_has_unicast_addresses (GstRTSPAddressPool * pool);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_ADDRESS_POOL_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-auth
+ * @short_description: Authentication and authorization
+ * @see_also: #GstRTSPPermissions, #GstRTSPToken
+ *
+ * The #GstRTSPAuth object is responsible for checking if the current user is
+ * allowed to perform requested actions. The default implementation has some
+ * reasonable checks but subclasses can implement custom security policies.
+ *
+ * A new auth object is made with gst_rtsp_auth_new(). It is usually configured
+ * on the #GstRTSPServer object.
+ *
+ * The RTSP server will call gst_rtsp_auth_check() with a string describing the
+ * check to perform. The possible checks are prefixed with
+ * GST_RTSP_AUTH_CHECK_*. Depending on the check, the default implementation
+ * will use the current #GstRTSPToken, #GstRTSPContext and
+ * #GstRTSPPermissions on the object to check if an operation is allowed.
+ *
+ * The default #GstRTSPAuth object has support for basic authentication. With
+ * gst_rtsp_auth_add_basic() you can add a basic authentication string together
+ * with the #GstRTSPToken that will become active when successfully
+ * authenticated.
+ *
+ * When a TLS certificate has been set with gst_rtsp_auth_set_tls_certificate(),
+ * the default auth object will require the client to connect with a TLS
+ * connection.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+
+#include <string.h>
+
+#include "rtsp-auth.h"
+
+#define GST_RTSP_AUTH_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_AUTH, GstRTSPAuthPrivate))
+
+struct _GstRTSPAuthPrivate
+{
+ GMutex lock;
+
+ /* the TLS certificate */
+ GTlsCertificate *certificate;
+ GHashTable *basic; /* protected by lock */
+ GstRTSPToken *default_token;
+ GstRTSPMethod methods;
+};
+
+enum
+{
+ PROP_0,
+ PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_auth_debug);
+#define GST_CAT_DEFAULT rtsp_auth_debug
+
+static void gst_rtsp_auth_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_auth_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_auth_finalize (GObject * obj);
+
+static gboolean default_authenticate (GstRTSPAuth * auth, GstRTSPContext * ctx);
+static gboolean default_check (GstRTSPAuth * auth, GstRTSPContext * ctx,
+ const gchar * check);
+
+G_DEFINE_TYPE (GstRTSPAuth, gst_rtsp_auth, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_auth_class_init (GstRTSPAuthClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPAuthPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_auth_get_property;
+ gobject_class->set_property = gst_rtsp_auth_set_property;
+ gobject_class->finalize = gst_rtsp_auth_finalize;
+
+ klass->authenticate = default_authenticate;
+ klass->check = default_check;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_auth_debug, "rtspauth", 0, "GstRTSPAuth");
+}
+
+static void
+gst_rtsp_auth_init (GstRTSPAuth * auth)
+{
+ GstRTSPAuthPrivate *priv;
+
+ auth->priv = priv = GST_RTSP_AUTH_GET_PRIVATE (auth);
+
+ g_mutex_init (&priv->lock);
+
+ priv->basic = g_hash_table_new_full (g_str_hash, g_str_equal, g_free,
+ (GDestroyNotify) gst_rtsp_token_unref);
+
+ /* bitwise or of all methods that need authentication */
+ priv->methods = 0;
+}
+
+static void
+gst_rtsp_auth_finalize (GObject * obj)
+{
+ GstRTSPAuth *auth = GST_RTSP_AUTH (obj);
+ GstRTSPAuthPrivate *priv = auth->priv;
+
+ GST_INFO ("finalize auth %p", auth);
+
+ if (priv->certificate)
+ g_object_unref (priv->certificate);
+ g_hash_table_unref (priv->basic);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_auth_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_auth_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ switch (propid) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_auth_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ switch (propid) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_auth_new:
+ *
+ * Create a new #GstRTSPAuth instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPAuth
+ */
+GstRTSPAuth *
+gst_rtsp_auth_new (void)
+{
+ GstRTSPAuth *result;
+
+ result = g_object_new (GST_TYPE_RTSP_AUTH, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_auth_set_tls_certificate:
+ * @auth: a #GstRTSPAuth
+ * @cert: (transfer none) (allow-none): a #GTlsCertificate
+ *
+ * Set the TLS certificate for the auth. Client connections will only
+ * be accepted when TLS is negotiated.
+ */
+void
+gst_rtsp_auth_set_tls_certificate (GstRTSPAuth * auth, GTlsCertificate * cert)
+{
+ GstRTSPAuthPrivate *priv;
+ GTlsCertificate *old;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+
+ priv = auth->priv;
+
+ if (cert)
+ g_object_ref (cert);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->certificate;
+ priv->certificate = cert;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_auth_get_tls_certificate:
+ * @auth: a #GstRTSPAuth
+ *
+ * Get the #GTlsCertificate used for negotiating TLS @auth.
+ *
+ * Returns: (transfer full): the #GTlsCertificate of @auth. g_object_unref() after
+ * usage.
+ */
+GTlsCertificate *
+gst_rtsp_auth_get_tls_certificate (GstRTSPAuth * auth)
+{
+ GstRTSPAuthPrivate *priv;
+ GTlsCertificate *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), NULL);
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->certificate))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_auth_set_default_token:
+ * @auth: a #GstRTSPAuth
+ * @token: (transfer none) (allow-none): a #GstRTSPToken
+ *
+ * Set the default #GstRTSPToken to @token in @auth. The default token will
+ * be used for unauthenticated users.
+ */
+void
+gst_rtsp_auth_set_default_token (GstRTSPAuth * auth, GstRTSPToken * token)
+{
+ GstRTSPAuthPrivate *priv;
+ GstRTSPToken *old;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+
+ priv = auth->priv;
+
+ if (token)
+ gst_rtsp_token_ref (token);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->default_token;
+ priv->default_token = token;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ gst_rtsp_token_unref (old);
+}
+
+/**
+ * gst_rtsp_auth_get_default_token:
+ * @auth: a #GstRTSPAuth
+ *
+ * Get the default token for @auth. This token will be used for unauthenticated
+ * users.
+ *
+ * Returns: (transfer full): the #GstRTSPToken of @auth. gst_rtsp_token_unref() after
+ * usage.
+ */
+GstRTSPToken *
+gst_rtsp_auth_get_default_token (GstRTSPAuth * auth)
+{
+ GstRTSPAuthPrivate *priv;
+ GstRTSPToken *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), NULL);
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->default_token))
+ gst_rtsp_token_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_auth_add_basic:
+ * @auth: a #GstRTSPAuth
+ * @basic: the basic token
+ * @token: (transfer none): authorisation token
+ *
+ * Add a basic token for the default authentication algorithm that
+ * enables the client with privileges listed in @token.
+ */
+void
+gst_rtsp_auth_add_basic (GstRTSPAuth * auth, const gchar * basic,
+ GstRTSPToken * token)
+{
+ GstRTSPAuthPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+ g_return_if_fail (basic != NULL);
+ g_return_if_fail (GST_IS_RTSP_TOKEN (token));
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_hash_table_replace (priv->basic, g_strdup (basic),
+ gst_rtsp_token_ref (token));
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_auth_remove_basic:
+ * @auth: a #GstRTSPAuth
+ * @basic: (transfer none): the basic token
+ *
+ * Add a basic token for the default authentication algorithm that
+ * enables the client with privileges from @authgroup.
+ */
+void
+gst_rtsp_auth_remove_basic (GstRTSPAuth * auth, const gchar * basic)
+{
+ GstRTSPAuthPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+ g_return_if_fail (basic != NULL);
+
+ priv = auth->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_hash_table_remove (priv->basic, basic);
+ g_mutex_unlock (&priv->lock);
+}
+
+static gboolean
+default_authenticate (GstRTSPAuth * auth, GstRTSPContext * ctx)
+{
+ GstRTSPAuthPrivate *priv = auth->priv;
+ GstRTSPResult res;
+ gchar *authorization;
+
+ GST_DEBUG_OBJECT (auth, "authenticate");
+
+ g_mutex_lock (&priv->lock);
+ /* FIXME, need to ref but we have no way to unref when the ctx is
+ * popped */
+ ctx->token = priv->default_token;
+ g_mutex_unlock (&priv->lock);
+
+ res =
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_AUTHORIZATION,
+ &authorization, 0);
+ if (res < 0)
+ goto no_auth;
+
+ /* parse type */
+ if (g_ascii_strncasecmp (authorization, "basic ", 6) == 0) {
+ GstRTSPToken *token;
+
+ GST_DEBUG_OBJECT (auth, "check Basic auth");
+ g_mutex_lock (&priv->lock);
+ if ((token = g_hash_table_lookup (priv->basic, &authorization[6]))) {
+ GST_DEBUG_OBJECT (auth, "setting token %p", token);
+ ctx->token = token;
+ }
+ g_mutex_unlock (&priv->lock);
+ } else if (g_ascii_strncasecmp (authorization, "digest ", 7) == 0) {
+ GST_DEBUG_OBJECT (auth, "check Digest auth");
+ /* not implemented yet */
+ }
+ return TRUE;
+
+no_auth:
+ {
+ GST_DEBUG_OBJECT (auth, "no authorization header found");
+ return TRUE;
+ }
+}
+
+static void
+send_response (GstRTSPAuth * auth, GstRTSPStatusCode code, GstRTSPContext * ctx)
+{
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ if (code == GST_RTSP_STS_UNAUTHORIZED) {
+ /* we only have Basic for now */
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_WWW_AUTHENTICATE,
+ "Basic realm=\"GStreamer RTSP Server\"");
+ }
+ gst_rtsp_client_send_message (ctx->client, ctx->session, ctx->response);
+}
+
+static gboolean
+ensure_authenticated (GstRTSPAuth * auth, GstRTSPContext * ctx)
+{
+ GstRTSPAuthClass *klass;
+
+ klass = GST_RTSP_AUTH_GET_CLASS (auth);
+
+ /* we need a token to check */
+ if (ctx->token == NULL) {
+ if (klass->authenticate) {
+ if (!klass->authenticate (auth, ctx))
+ goto authenticate_failed;
+ }
+ }
+ if (ctx->token == NULL)
+ goto no_auth;
+
+ return TRUE;
+
+/* ERRORS */
+authenticate_failed:
+ {
+ GST_DEBUG_OBJECT (auth, "authenticate failed");
+ send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+ return FALSE;
+ }
+no_auth:
+ {
+ GST_DEBUG_OBJECT (auth, "no authorization token found");
+ send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+ return FALSE;
+ }
+}
+
+/* new connection */
+static gboolean
+check_connect (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+ GstRTSPAuthPrivate *priv = auth->priv;
+
+ if (priv->certificate) {
+ GTlsConnection *tls;
+
+ /* configure the connection */
+ tls = gst_rtsp_connection_get_tls (ctx->conn, NULL);
+ g_tls_connection_set_certificate (tls, priv->certificate);
+ }
+ return TRUE;
+}
+
+/* check url and methods */
+static gboolean
+check_url (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+ GstRTSPAuthPrivate *priv = auth->priv;
+
+ if ((ctx->method & priv->methods) != 0)
+ if (!ensure_authenticated (auth, ctx))
+ goto not_authenticated;
+
+ return TRUE;
+
+ /* ERRORS */
+not_authenticated:
+ {
+ return FALSE;
+ }
+}
+
+/* check access to media factory */
+static gboolean
+check_factory (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+ const gchar *role;
+ GstRTSPPermissions *perms;
+
+ if (!ensure_authenticated (auth, ctx))
+ return FALSE;
+
+ if (!(role = gst_rtsp_token_get_string (ctx->token,
+ GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE)))
+ goto no_media_role;
+ if (!(perms = gst_rtsp_media_factory_get_permissions (ctx->factory)))
+ goto no_permissions;
+
+ if (g_str_equal (check, GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS)) {
+ if (!gst_rtsp_permissions_is_allowed (perms, role,
+ GST_RTSP_PERM_MEDIA_FACTORY_ACCESS))
+ goto no_access;
+ } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT)) {
+ if (!gst_rtsp_permissions_is_allowed (perms, role,
+ GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT))
+ goto no_construct;
+ }
+
+ gst_rtsp_permissions_unref (perms);
+
+ return TRUE;
+
+ /* ERRORS */
+no_media_role:
+ {
+ GST_DEBUG_OBJECT (auth, "no media factory role found");
+ send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+ return FALSE;
+ }
+no_permissions:
+ {
+ GST_DEBUG_OBJECT (auth, "no permissions on media factory found");
+ send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+ return FALSE;
+ }
+no_access:
+ {
+ GST_DEBUG_OBJECT (auth, "no permissions to access media factory");
+ gst_rtsp_permissions_unref (perms);
+ send_response (auth, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_construct:
+ {
+ GST_DEBUG_OBJECT (auth, "no permissions to construct media factory");
+ gst_rtsp_permissions_unref (perms);
+ send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+ return FALSE;
+ }
+}
+
+static gboolean
+check_client_settings (GstRTSPAuth * auth, GstRTSPContext * ctx,
+ const gchar * check)
+{
+ if (!ensure_authenticated (auth, ctx))
+ return FALSE;
+
+ return gst_rtsp_token_is_allowed (ctx->token,
+ GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS);
+}
+
+static gboolean
+default_check (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+ gboolean res = FALSE;
+
+ /* FIXME, use hastable or so */
+ if (g_str_equal (check, GST_RTSP_AUTH_CHECK_CONNECT)) {
+ res = check_connect (auth, ctx, check);
+ } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_URL)) {
+ res = check_url (auth, ctx, check);
+ } else if (g_str_has_prefix (check, "auth.check.media.factory.")) {
+ res = check_factory (auth, ctx, check);
+ } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS)) {
+ res = check_client_settings (auth, ctx, check);
+ }
+ return res;
+}
+
+static gboolean
+no_auth_check (const gchar * check)
+{
+ gboolean res;
+
+ if (g_str_equal (check, GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS))
+ res = FALSE;
+ else
+ res = TRUE;
+
+ return res;
+}
+
+/**
+ * gst_rtsp_auth_check:
+ * @check: the item to check
+ *
+ * Check if @check is allowed in the current context.
+ *
+ * Returns: FALSE if check failed.
+ */
+gboolean
+gst_rtsp_auth_check (const gchar * check)
+{
+ gboolean result = FALSE;
+ GstRTSPAuthClass *klass;
+ GstRTSPContext *ctx;
+ GstRTSPAuth *auth;
+
+ g_return_val_if_fail (check != NULL, FALSE);
+
+ if (!(ctx = gst_rtsp_context_get_current ()))
+ goto no_context;
+
+ /* no auth, we don't need to check */
+ if (!(auth = ctx->auth))
+ return no_auth_check (check);
+
+ klass = GST_RTSP_AUTH_GET_CLASS (auth);
+
+ GST_DEBUG_OBJECT (auth, "check authorization '%s'", check);
+
+ if (klass->check)
+ result = klass->check (auth, ctx, check);
+
+ return result;
+
+ /* ERRORS */
+no_context:
+ {
+ GST_ERROR ("no context found");
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_auth_make_basic:
+ * @user: a userid
+ * @pass: a password
+ *
+ * Construct a Basic authorisation token from @user and @pass.
+ *
+ * Returns: (transfer full): the base64 encoding of the string @user:@pass.
+ * g_free() after usage.
+ */
+gchar *
+gst_rtsp_auth_make_basic (const gchar * user, const gchar * pass)
+{
+ gchar *user_pass;
+ gchar *result;
+
+ g_return_val_if_fail (user != NULL, NULL);
+ g_return_val_if_fail (pass != NULL, NULL);
+
+ user_pass = g_strjoin (":", user, pass, NULL);
+ result = g_base64_encode ((guchar *) user_pass, strlen (user_pass));
+ g_free (user_pass);
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_AUTH_H__
+#define __GST_RTSP_AUTH_H__
+
+typedef struct _GstRTSPAuth GstRTSPAuth;
+typedef struct _GstRTSPAuthClass GstRTSPAuthClass;
+typedef struct _GstRTSPAuthPrivate GstRTSPAuthPrivate;
+
+#include "rtsp-client.h"
+#include "rtsp-token.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_AUTH (gst_rtsp_auth_get_type ())
+#define GST_IS_RTSP_AUTH(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_AUTH))
+#define GST_IS_RTSP_AUTH_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_AUTH))
+#define GST_RTSP_AUTH_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_AUTH, GstRTSPAuthClass))
+#define GST_RTSP_AUTH(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_AUTH, GstRTSPAuth))
+#define GST_RTSP_AUTH_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_AUTH, GstRTSPAuthClass))
+#define GST_RTSP_AUTH_CAST(obj) ((GstRTSPAuth*)(obj))
+#define GST_RTSP_AUTH_CLASS_CAST(klass) ((GstRTSPAuthClass*)(klass))
+
+/**
+ * GstRTSPAuth:
+ *
+ * The authentication structure.
+ */
+struct _GstRTSPAuth {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPAuthPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPAuthClass:
+ * @authenticate: check the authentication of a client. The default implementation
+ * checks if the authentication in the header matches one of the basic
+ * authentication tokens. This function should set the authgroup field
+ * in the context.
+ * @check: check if a resource can be accessed. this function should
+ * call authenticate to authenticate the client when needed. The method
+ * should also construct and send an appropriate response message on
+ * error.
+ *
+ * The authentication class.
+ */
+struct _GstRTSPAuthClass {
+ GObjectClass parent_class;
+
+ gboolean (*authenticate) (GstRTSPAuth *auth, GstRTSPContext *ctx);
+ gboolean (*check) (GstRTSPAuth *auth, GstRTSPContext *ctx,
+ const gchar *check);
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_auth_get_type (void);
+
+GstRTSPAuth * gst_rtsp_auth_new (void);
+
+void gst_rtsp_auth_set_tls_certificate (GstRTSPAuth *auth, GTlsCertificate *cert);
+GTlsCertificate * gst_rtsp_auth_get_tls_certificate (GstRTSPAuth *auth);
+
+void gst_rtsp_auth_set_default_token (GstRTSPAuth *auth, GstRTSPToken *token);
+GstRTSPToken * gst_rtsp_auth_get_default_token (GstRTSPAuth *auth);
+
+void gst_rtsp_auth_add_basic (GstRTSPAuth *auth, const gchar * basic,
+ GstRTSPToken *token);
+void gst_rtsp_auth_remove_basic (GstRTSPAuth *auth, const gchar * basic);
+
+gboolean gst_rtsp_auth_check (const gchar *check);
+
+
+/* helpers */
+gchar * gst_rtsp_auth_make_basic (const gchar * user, const gchar * pass);
+
+/* checks */
+/**
+ * GST_RTSP_AUTH_CHECK_CONNECT:
+ *
+ * Check a new connection
+ */
+#define GST_RTSP_AUTH_CHECK_CONNECT "auth.check.connect"
+/**
+ * GST_RTSP_AUTH_CHECK_URL:
+ *
+ * Check the URL and methods
+ */
+#define GST_RTSP_AUTH_CHECK_URL "auth.check.url"
+/**
+ * GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS:
+ *
+ * Check if access is allowed to a factory.
+ * When access is not allowed an 404 Not Found is sent in the response.
+ */
+#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS "auth.check.media.factory.access"
+/**
+ * GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT:
+ *
+ * Check if media can be constructed from a media factory
+ * A response should be sent on error.
+ */
+#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT "auth.check.media.factory.construct"
+/**
+ * GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS:
+ *
+ * Check if the client can specify TTL, destination and
+ * port pair in multicast. No response is sent when the check returns
+ * %FALSE.
+ */
+#define GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS "auth.check.transport.client-settings"
+
+
+/* tokens */
+/**
+ * GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE:
+ *
+ * G_TYPE_STRING, the role to use when dealing with media factories
+ *
+ * The default #GstRTSPAuth object uses this string in the token to find the
+ * role of the media factory. It will then retrieve the #GstRTSPPermissions of
+ * the media factory and retrieve the role with the same name.
+ */
+#define GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE "media.factory.role"
+/**
+ * GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS:
+ *
+ * G_TYPE_BOOLEAN, %TRUE if the client can specify TTL, destination and
+ * port pair in multicast.
+ */
+#define GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS "transport.client-settings"
+
+/* permissions */
+/**
+ * GST_RTSP_PERM_MEDIA_FACTORY_ACCESS:
+ *
+ * G_TYPE_BOOLEAN, %TRUE if the media can be accessed, %FALSE will
+ * return a 404 Not Found error when trying to access the media.
+ */
+#define GST_RTSP_PERM_MEDIA_FACTORY_ACCESS "media.factory.access"
+/**
+ * GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT:
+ *
+ * G_TYPE_BOOLEAN, %TRUE if the media can be constructed, %FALSE will
+ * return a 404 Not Found error when trying to access the media.
+ */
+#define GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT "media.factory.construct"
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_AUTH_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-client
+ * @short_description: A client connection state
+ * @see_also: #GstRTSPServer, #GstRTSPThreadPool
+ *
+ * The client object handles the connection with a client for as long as a TCP
+ * connection is open.
+ *
+ * A #GstRTSPWFDClient is created by #GstRTSPServer when a new connection is
+ * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
+ * #GstRTSPAuth and #GstRTSPThreadPool from the server.
+ *
+ * The client connection should be configured with the #GstRTSPConnection using
+ * gst_rtsp_wfd_client_set_connection() before it can be attached to a #GMainContext
+ * using gst_rtsp_wfd_client_attach(). From then on the client will handle requests
+ * on the connection.
+ *
+ * Use gst_rtsp_wfd_client_session_filter() to iterate or modify all the
+ * #GstRTSPSession objects managed by the client object.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include <stdio.h>
+#include <string.h>
+
+#include "rtsp-client-wfd.h"
+#include "rtsp-media-factory-wfd.h"
+#include "rtsp-sdp.h"
+#include "rtsp-params.h"
+
+#define GST_RTSP_WFD_CLIENT_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_WFD_CLIENT, GstRTSPWFDClientPrivate))
+
+typedef struct _GstRTSPClientRTPStats GstRTSPClientRTPStats;
+
+struct _GstRTSPClientRTPStats {
+ GstRTSPStream *stream;
+ guint64 last_sent_bytes;
+ guint64 sent_bytes;
+ guint last_seqnum;
+ guint seqnum;
+
+ /* Info in RR (Receiver Report) */
+ guint8 fraction_lost;
+ guint32 cumulative_lost_num;
+ guint16 max_seqnum;
+ guint32 arrival_jitter;
+ guint32 lsr;
+ guint32 dlsr;
+ guint32 rtt;
+ guint resent_packets;
+};
+
+struct _GstRTSPWFDClientPrivate
+{
+ GstRTSPWFDClientSendFunc send_func; /* protected by send_lock */
+ gpointer send_data; /* protected by send_lock */
+ GDestroyNotify send_notify; /* protected by send_lock */
+
+ /* used to cache the media in the last requested DESCRIBE so that
+ * we can pick it up in the next SETUP immediately */
+ gchar *path;
+ GstRTSPMedia *media;
+
+ GList *transports;
+ GList *sessions;
+
+ guint8 m1_done;
+ guint8 m3_done;
+ guint8 m4_done;
+
+ /* Host's URL info */
+ gchar *host_address;
+
+ /* Parameters for WIFI-DISPLAY */
+ guint caCodec;
+ guint8 audio_codec;
+ guint cFreq;
+ guint cChanels;
+ guint cBitwidth;
+ guint caLatency;
+ guint cvCodec;
+ guint cNative;
+ guint64 cNativeResolution;
+ guint64 video_resolution_supported;
+ gint video_native_resolution;
+ guint64 cCEAResolution;
+ guint64 cVESAResolution;
+ guint64 cHHResolution;
+ guint cProfile;
+ guint cLevel;
+ guint32 cMaxHeight;
+ guint32 cMaxWidth;
+ guint32 cFramerate;
+ guint32 cInterleaved;
+ guint32 cmin_slice_size;
+ guint32 cslice_enc_params;
+ guint cframe_rate_control;
+ guint cvLatency;
+ guint ctrans;
+ guint cprofile;
+ guint clowertrans;
+ guint32 crtp_port0;
+ guint32 crtp_port1;
+
+ gboolean protection_enabled;
+ GstWFDHDCPProtection hdcp_version;
+ guint32 hdcp_tcpport;
+
+ gboolean edid_supported;
+ guint32 edid_hres;
+ guint32 edid_vres;
+
+ gboolean keep_alive_flag;
+ GMutex keep_alive_lock;
+
+ /* RTP statistics */
+ GstRTSPClientRTPStats stats;
+ GMutex stats_lock;
+ guint stats_timer_id;
+ gboolean rtcp_stats_enabled;
+};
+
+#define DEFAULT_WFD_TIMEOUT 60
+#define WFD_MOUNT_POINT "/wfd1.0/streamid=0"
+
+enum
+{
+ SIGNAL_WFD_OPTIONS_REQUEST,
+ SIGNAL_WFD_GET_PARAMETER_REQUEST,
+ SIGNAL_WFD_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_wfd_client_debug);
+#define GST_CAT_DEFAULT rtsp_wfd_client_debug
+
+static guint gst_rtsp_client_wfd_signals[SIGNAL_WFD_LAST] = { 0 };
+
+static void gst_rtsp_wfd_client_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_wfd_client_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_wfd_client_finalize (GObject * obj);
+
+static gboolean handle_wfd_options_request (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static gboolean handle_wfd_set_param_request (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static gboolean handle_wfd_get_param_request (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+
+static void send_generic_wfd_response (GstRTSPWFDClient * client,
+ GstRTSPStatusCode code, GstRTSPContext * ctx);
+static gchar *wfd_make_path_from_uri (GstRTSPClient * client,
+ const GstRTSPUrl * uri);
+static void wfd_options_request_done (GstRTSPWFDClient * client, GstRTSPContext *ctx);
+static void wfd_get_param_request_done (GstRTSPWFDClient * client, GstRTSPContext *ctx);
+static void handle_wfd_response (GstRTSPClient * client, GstRTSPContext * ctx);
+static void handle_wfd_play (GstRTSPClient * client, GstRTSPContext * ctx);
+static void wfd_set_keep_alive_condition(GstRTSPWFDClient * client);
+static gboolean wfd_ckeck_keep_alive_response (gpointer userdata);
+static gboolean keep_alive_condition(gpointer userdata);
+static gboolean wfd_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+ GstRTSPStream * stream, GstRTSPContext * ctx);
+
+GstRTSPResult prepare_trigger_request (GstRTSPWFDClient * client,
+ GstRTSPMessage * request, GstWFDTriggerType trigger_type, gchar * url);
+
+GstRTSPResult prepare_request (GstRTSPWFDClient * client,
+ GstRTSPMessage * request, GstRTSPMethod method, gchar * url);
+
+void
+send_request (GstRTSPWFDClient * client, GstRTSPSession * session,
+ GstRTSPMessage * request);
+
+GstRTSPResult
+prepare_response (GstRTSPWFDClient * client, GstRTSPMessage * request,
+ GstRTSPMessage * response, GstRTSPMethod method);
+
+static GstRTSPResult handle_M1_message (GstRTSPWFDClient * client);
+static GstRTSPResult handle_M3_message (GstRTSPWFDClient * client);
+static GstRTSPResult handle_M4_message (GstRTSPWFDClient * client);
+static GstRTSPResult handle_M16_message (GstRTSPWFDClient * client);
+
+G_DEFINE_TYPE (GstRTSPWFDClient, gst_rtsp_wfd_client, GST_TYPE_RTSP_CLIENT);
+
+static void
+gst_rtsp_wfd_client_class_init (GstRTSPWFDClientClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstRTSPClientClass *rtsp_client_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPWFDClientPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ rtsp_client_class = GST_RTSP_CLIENT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_wfd_client_get_property;
+ gobject_class->set_property = gst_rtsp_wfd_client_set_property;
+ gobject_class->finalize = gst_rtsp_wfd_client_finalize;
+
+ //klass->create_sdp = create_sdp;
+ //klass->configure_client_transport = default_configure_client_transport;
+ //klass->params_set = default_params_set;
+ //klass->params_get = default_params_get;
+
+ rtsp_client_class->handle_options_request = handle_wfd_options_request;
+ rtsp_client_class->handle_set_param_request = handle_wfd_set_param_request;
+ rtsp_client_class->handle_get_param_request = handle_wfd_get_param_request;
+ rtsp_client_class->make_path_from_uri = wfd_make_path_from_uri;
+ rtsp_client_class->configure_client_media = wfd_configure_client_media;
+
+ rtsp_client_class->handle_response = handle_wfd_response;
+ rtsp_client_class->play_request = handle_wfd_play;
+
+ gst_rtsp_client_wfd_signals[SIGNAL_WFD_OPTIONS_REQUEST] =
+ g_signal_new ("wfd-options-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPWFDClientClass,
+ wfd_options_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ gst_rtsp_client_wfd_signals[SIGNAL_WFD_GET_PARAMETER_REQUEST] =
+ g_signal_new ("wfd-get-parameter-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPWFDClientClass,
+ wfd_get_param_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ klass->wfd_options_request = wfd_options_request_done;
+ klass->wfd_get_param_request = wfd_get_param_request_done;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_wfd_client_debug, "rtspwfdclient", 0,
+ "GstRTSPWFDClient");
+}
+
+static void
+gst_rtsp_wfd_client_init (GstRTSPWFDClient * client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ g_return_if_fail (priv != NULL);
+
+ client->priv = priv;
+ priv->protection_enabled = FALSE;
+ priv->video_native_resolution = GST_WFD_VIDEO_CEA_RESOLUTION;
+ priv->video_resolution_supported = GST_WFD_CEA_640x480P60;
+ priv->audio_codec = GST_WFD_AUDIO_AAC;
+ priv->keep_alive_flag = FALSE;
+
+ g_mutex_init (&priv->keep_alive_lock);
+ g_mutex_init (&priv->stats_lock);
+
+ priv->host_address = NULL;
+
+ priv->stats_timer_id = -1;
+ priv->rtcp_stats_enabled = FALSE;
+ memset (&priv->stats, 0x00, sizeof (GstRTSPClientRTPStats));
+
+ GST_INFO_OBJECT (client, "Client is initialized");
+}
+
+/* A client is finalized when the connection is broken */
+static void
+gst_rtsp_wfd_client_finalize (GObject * obj)
+{
+ GstRTSPWFDClient *client = GST_RTSP_WFD_CLIENT (obj);
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ g_return_if_fail (GST_IS_RTSP_WFD_CLIENT (obj));
+ g_return_if_fail (priv != NULL);
+
+ GST_INFO ("finalize client %p", client);
+
+ if (priv->host_address)
+ g_free (priv->host_address);
+
+ if (priv->stats_timer_id > 0)
+ g_source_remove(priv->stats_timer_id);
+
+ g_mutex_clear (&priv->keep_alive_lock);
+ g_mutex_clear (&priv->stats_lock);
+ G_OBJECT_CLASS (gst_rtsp_wfd_client_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_wfd_client_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ //GstRTSPWFDClient *client = GST_RTSP_WFD_CLIENT (object);
+
+ switch (propid) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_wfd_client_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ //GstRTSPWFDClient *client = GST_RTSP_WFD_CLIENT (object);
+
+ switch (propid) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_wfd_client_new:
+ *
+ * Create a new #GstRTSPWFDClient instance.
+ *
+ * Returns: a new #GstRTSPWFDClient
+ */
+GstRTSPWFDClient *
+gst_rtsp_wfd_client_new (void)
+{
+ GstRTSPWFDClient *result;
+
+ result = g_object_new (GST_TYPE_RTSP_WFD_CLIENT, NULL);
+
+ return result;
+}
+
+void
+gst_rtsp_wfd_client_start_wfd (GstRTSPWFDClient * client)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GST_INFO_OBJECT (client, "gst_rtsp_wfd_client_start_wfd");
+
+ res = handle_M1_message (client);
+ if (res < GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "handle_M1_message failed : %d", res);
+ }
+
+ return;
+}
+
+static gboolean
+wfd_display_rtp_stats (gpointer userdata)
+{
+ guint16 seqnum = 0;
+ guint64 bytes = 0;
+
+ GstRTSPWFDClient *client = NULL;
+ GstRTSPWFDClientPrivate *priv = NULL;
+
+ client = (GstRTSPWFDClient *) userdata;
+ priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ if (!priv) {
+ GST_ERROR("No priv");
+ return FALSE;
+ }
+
+ g_mutex_lock(&priv->stats_lock);
+
+ seqnum = gst_rtsp_stream_get_current_seqnum (priv->stats.stream);
+ bytes = gst_rtsp_stream_get_udp_sent_bytes (priv->stats.stream);
+
+ GST_INFO ("----------------------------------------------------\n");
+ GST_INFO ("Sent RTP packets : %d", seqnum - priv->stats.last_seqnum);
+ GST_INFO ("Sent Bytes of RTP packets : %lld bytes", bytes - priv->stats.last_sent_bytes);
+
+ priv->stats.last_seqnum = seqnum;
+ priv->stats.last_sent_bytes = bytes;
+
+ if (priv->rtcp_stats_enabled) {
+ GST_INFO ("Fraction Lost: %d", priv->stats.fraction_lost);
+ GST_INFO ("Cumulative number of packets lost: %d", priv->stats.cumulative_lost_num);
+ GST_INFO ("Extended highest sequence number received: %d", priv->stats.max_seqnum);
+ GST_INFO ("Interarrival Jitter: %d", priv->stats.arrival_jitter);
+ GST_INFO ("Round trip time : %d", priv->stats.rtt);
+ }
+
+ GST_INFO ("----------------------------------------------------\n");
+
+ g_mutex_unlock(&priv->stats_lock);
+
+ return TRUE;
+}
+
+static void
+on_rtcp_stats (GstRTSPStream *stream, GstStructure *stats, GstRTSPClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ guint fraction_lost, exthighestseq, jitter, lsr, dlsr, rtt;
+ gint packetslost;
+
+ if (!priv) return;
+
+ g_mutex_lock(&priv->stats_lock);
+
+ gst_structure_get_uint (stats, "rb-fractionlost", &fraction_lost);
+ gst_structure_get_int (stats, "rb-packetslost", &packetslost);
+ gst_structure_get_uint (stats, "rb-exthighestseq", &exthighestseq);
+ gst_structure_get_uint (stats, "rb-jitter", &jitter);
+ gst_structure_get_uint (stats, "rb-lsr", &lsr);
+ gst_structure_get_uint (stats, "rb-dlsr", &dlsr);
+ gst_structure_get_uint (stats, "rb-round-trip", &rtt);
+
+ if (!priv->rtcp_stats_enabled)
+ priv->rtcp_stats_enabled = TRUE;
+
+ priv->stats.stream = stream;
+ priv->stats.fraction_lost = (guint8)fraction_lost;
+ priv->stats.cumulative_lost_num += (guint32)fraction_lost;
+ priv->stats.max_seqnum = (guint16)exthighestseq;
+ priv->stats.arrival_jitter = (guint32)jitter;
+ priv->stats.lsr = (guint32)lsr;
+ priv->stats.dlsr = (guint32)dlsr;
+ priv->stats.rtt = (guint32)rtt;
+
+ g_mutex_unlock(&priv->stats_lock);
+}
+
+static gboolean
+wfd_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+ GstRTSPStream * stream, GstRTSPContext * ctx)
+{
+ if (stream) {
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ if (priv)
+ priv->stats.stream = stream;
+ g_signal_connect (stream, "rtcp-statistics", (GCallback) on_rtcp_stats, client);
+ }
+
+ return GST_RTSP_CLIENT_CLASS (gst_rtsp_wfd_client_parent_class)->configure_client_media (client, media, stream, ctx);
+}
+static void
+wfd_options_request_done (GstRTSPWFDClient * client, GstRTSPContext *ctx)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPWFDClientClass *klass = GST_RTSP_WFD_CLIENT_GET_CLASS (client);
+
+ g_return_if_fail (klass != NULL);
+
+ GST_INFO_OBJECT (client, "M2 done..");
+
+ res = handle_M3_message (client);
+ if (res < GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "handle_M3_message failed : %d", res);
+ }
+
+ if (klass->prepare_resource) {
+ klass->prepare_resource (client, ctx);
+ }
+
+ return;
+}
+
+static void
+wfd_get_param_request_done (GstRTSPWFDClient * client, GstRTSPContext *ctx)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ GstRTSPWFDClientClass *klass = GST_RTSP_WFD_CLIENT_GET_CLASS (client);
+
+ g_return_if_fail (priv != NULL && klass != NULL);
+
+ priv->m3_done = TRUE;
+ GST_INFO_OBJECT (client, "M3 done..");
+
+ res = handle_M4_message (client);
+ if (res < GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "handle_M4_message failed : %d", res);
+ }
+
+ if (klass->confirm_resource) {
+ klass->confirm_resource (client, ctx);
+ }
+
+ return;
+}
+
+static guint
+wfd_get_prefered_audio_codec (guint8 srcAudioCodec,
+ guint sinkAudioCodec)
+{
+ int i = 0;
+ guint codec = 0;
+ for (i = 0; i < 8; i++) {
+ if (((sinkAudioCodec << i) & 0x80)
+ && ((srcAudioCodec << i) & 0x80)) {
+ codec = (0x01 << (7 - i));
+ break;
+ }
+ }
+ return codec;
+}
+
+static guint64
+wfd_get_prefered_resolution (guint64 srcResolution,
+ guint64 sinkResolution,
+ GstWFDVideoNativeResolution native,
+ guint32 * cMaxWidth,
+ guint32 * cMaxHeight, guint32 * cFramerate, guint32 * interleaved)
+{
+ int i = 0;
+ guint64 resolution = 0;
+ for (i = 0; i < 32; i++) {
+ if (((sinkResolution << i) & 0x80000000)
+ && ((srcResolution << i) & 0x80000000)) {
+ resolution = ((guint64) 0x00000001 << (31 - i));
+ break;
+ }
+ }
+ switch (native) {
+ case GST_WFD_VIDEO_CEA_RESOLUTION:
+ {
+ switch (resolution) {
+ case GST_WFD_CEA_640x480P60:
+ *cMaxWidth = 640;
+ *cMaxHeight = 480;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_720x480P60:
+ *cMaxWidth = 720;
+ *cMaxHeight = 480;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_720x480I60:
+ *cMaxWidth = 720;
+ *cMaxHeight = 480;
+ *cFramerate = 60;
+ *interleaved = 1;
+ break;
+ case GST_WFD_CEA_720x576P50:
+ *cMaxWidth = 720;
+ *cMaxHeight = 576;
+ *cFramerate = 50;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_720x576I50:
+ *cMaxWidth = 720;
+ *cMaxHeight = 576;
+ *cFramerate = 50;
+ *interleaved = 1;
+ break;
+ case GST_WFD_CEA_1280x720P30:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 720;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_1280x720P60:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 720;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_1920x1080P30:
+ *cMaxWidth = 1920;
+ *cMaxHeight = 1080;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_1920x1080P60:
+ *cMaxWidth = 1920;
+ *cMaxHeight = 1080;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_1920x1080I60:
+ *cMaxWidth = 1920;
+ *cMaxHeight = 1080;
+ *cFramerate = 60;
+ *interleaved = 1;
+ break;
+ case GST_WFD_CEA_1280x720P25:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 720;
+ *cFramerate = 25;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_1280x720P50:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 720;
+ *cFramerate = 50;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_1920x1080P25:
+ *cMaxWidth = 1920;
+ *cMaxHeight = 1080;
+ *cFramerate = 25;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_1920x1080P50:
+ *cMaxWidth = 1920;
+ *cMaxHeight = 1080;
+ *cFramerate = 50;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_1920x1080I50:
+ *cMaxWidth = 1920;
+ *cMaxHeight = 1080;
+ *cFramerate = 50;
+ *interleaved = 1;
+ break;
+ case GST_WFD_CEA_1280x720P24:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 720;
+ *cFramerate = 24;
+ *interleaved = 0;
+ break;
+ case GST_WFD_CEA_1920x1080P24:
+ *cMaxWidth = 1920;
+ *cMaxHeight = 1080;
+ *cFramerate = 24;
+ *interleaved = 0;
+ break;
+ default:
+ *cMaxWidth = 0;
+ *cMaxHeight = 0;
+ *cFramerate = 0;
+ *interleaved = 0;
+ break;
+ }
+ }
+ break;
+ case GST_WFD_VIDEO_VESA_RESOLUTION:
+ {
+ switch (resolution) {
+ case GST_WFD_VESA_800x600P30:
+ *cMaxWidth = 800;
+ *cMaxHeight = 600;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_800x600P60:
+ *cMaxWidth = 800;
+ *cMaxHeight = 600;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1024x768P30:
+ *cMaxWidth = 1024;
+ *cMaxHeight = 768;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1024x768P60:
+ *cMaxWidth = 1024;
+ *cMaxHeight = 768;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1152x864P30:
+ *cMaxWidth = 1152;
+ *cMaxHeight = 864;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1152x864P60:
+ *cMaxWidth = 1152;
+ *cMaxHeight = 864;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1280x768P30:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 768;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1280x768P60:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 768;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1280x800P30:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 800;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1280x800P60:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 800;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1360x768P30:
+ *cMaxWidth = 1360;
+ *cMaxHeight = 768;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1360x768P60:
+ *cMaxWidth = 1360;
+ *cMaxHeight = 768;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1366x768P30:
+ *cMaxWidth = 1366;
+ *cMaxHeight = 768;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1366x768P60:
+ *cMaxWidth = 1366;
+ *cMaxHeight = 768;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1280x1024P30:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 1024;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1280x1024P60:
+ *cMaxWidth = 1280;
+ *cMaxHeight = 1024;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1400x1050P30:
+ *cMaxWidth = 1400;
+ *cMaxHeight = 1050;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1400x1050P60:
+ *cMaxWidth = 1400;
+ *cMaxHeight = 1050;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1440x900P30:
+ *cMaxWidth = 1440;
+ *cMaxHeight = 900;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1440x900P60:
+ *cMaxWidth = 1440;
+ *cMaxHeight = 900;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1600x900P30:
+ *cMaxWidth = 1600;
+ *cMaxHeight = 900;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1600x900P60:
+ *cMaxWidth = 1600;
+ *cMaxHeight = 900;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1600x1200P30:
+ *cMaxWidth = 1600;
+ *cMaxHeight = 1200;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1600x1200P60:
+ *cMaxWidth = 1600;
+ *cMaxHeight = 1200;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1680x1024P30:
+ *cMaxWidth = 1680;
+ *cMaxHeight = 1024;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1680x1024P60:
+ *cMaxWidth = 1680;
+ *cMaxHeight = 1024;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1680x1050P30:
+ *cMaxWidth = 1680;
+ *cMaxHeight = 1050;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1680x1050P60:
+ *cMaxWidth = 1680;
+ *cMaxHeight = 1050;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1920x1200P30:
+ *cMaxWidth = 1920;
+ *cMaxHeight = 1200;
+ *cFramerate = 30;
+ *interleaved = 0;
+ break;
+ case GST_WFD_VESA_1920x1200P60:
+ *cMaxWidth = 1920;
+ *cMaxHeight = 1200;
+ *cFramerate = 60;
+ *interleaved = 0;
+ break;
+ default:
+ *cMaxWidth = 0;
+ *cMaxHeight = 0;
+ *cFramerate = 0;
+ *interleaved = 0;
+ break;
+ }
+ }
+ break;
+ case GST_WFD_VIDEO_HH_RESOLUTION:
+ {
+ *interleaved = 0;
+ switch (resolution) {
+ case GST_WFD_HH_800x480P30:
+ *cMaxWidth = 800;
+ *cMaxHeight = 480;
+ *cFramerate = 30;
+ break;
+ case GST_WFD_HH_800x480P60:
+ *cMaxWidth = 800;
+ *cMaxHeight = 480;
+ *cFramerate = 60;
+ break;
+ case GST_WFD_HH_854x480P30:
+ *cMaxWidth = 854;
+ *cMaxHeight = 480;
+ *cFramerate = 30;
+ break;
+ case GST_WFD_HH_854x480P60:
+ *cMaxWidth = 854;
+ *cMaxHeight = 480;
+ *cFramerate = 60;
+ break;
+ case GST_WFD_HH_864x480P30:
+ *cMaxWidth = 864;
+ *cMaxHeight = 480;
+ *cFramerate = 30;
+ break;
+ case GST_WFD_HH_864x480P60:
+ *cMaxWidth = 864;
+ *cMaxHeight = 480;
+ *cFramerate = 60;
+ break;
+ case GST_WFD_HH_640x360P30:
+ *cMaxWidth = 640;
+ *cMaxHeight = 360;
+ *cFramerate = 30;
+ break;
+ case GST_WFD_HH_640x360P60:
+ *cMaxWidth = 640;
+ *cMaxHeight = 360;
+ *cFramerate = 60;
+ break;
+ case GST_WFD_HH_960x540P30:
+ *cMaxWidth = 960;
+ *cMaxHeight = 540;
+ *cFramerate = 30;
+ break;
+ case GST_WFD_HH_960x540P60:
+ *cMaxWidth = 960;
+ *cMaxHeight = 540;
+ *cFramerate = 60;
+ break;
+ case GST_WFD_HH_848x480P30:
+ *cMaxWidth = 848;
+ *cMaxHeight = 480;
+ *cFramerate = 30;
+ break;
+ case GST_WFD_HH_848x480P60:
+ *cMaxWidth = 848;
+ *cMaxHeight = 480;
+ *cFramerate = 60;
+ break;
+ default:
+ *cMaxWidth = 0;
+ *cMaxHeight = 0;
+ *cFramerate = 0;
+ *interleaved = 0;
+ break;
+ }
+ }
+ break;
+
+ default:
+ *cMaxWidth = 0;
+ *cMaxHeight = 0;
+ *cFramerate = 0;
+ *interleaved = 0;
+ break;
+ }
+ return resolution;
+}
+
+static gchar *
+wfd_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
+{
+ gchar *path;
+
+ GST_DEBUG_OBJECT (client, "Got URI host : %s", uri->host);
+ GST_DEBUG_OBJECT (client, "Got URI abspath : %s", uri->abspath);
+
+ path = g_strdup ("/wfd1.0/streamid=0");
+
+ return path;
+}
+
+static void
+handle_wfd_play (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPWFDClient *_client = GST_RTSP_WFD_CLIENT (client);
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ g_return_if_fail (priv != NULL);
+
+ wfd_set_keep_alive_condition(_client);
+
+ priv->stats_timer_id = g_timeout_add (2000, wfd_display_rtp_stats, _client);
+}
+
+static void
+handle_wfd_response (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ guint8 *data = NULL;
+ guint size = 0;
+
+ GstRTSPWFDClient *_client = GST_RTSP_WFD_CLIENT (client);
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ g_return_if_fail (priv != NULL);
+
+ GST_INFO_OBJECT (_client, "Handling response..");
+
+ if (!ctx) {
+ GST_ERROR_OBJECT (_client, "Context is NULL");
+ goto error;
+ }
+
+ if (!ctx->response) {
+ GST_ERROR_OBJECT (_client, "Response is NULL");
+ goto error;
+ }
+
+ /* parsing the GET_PARAMTER response */
+ res = gst_rtsp_message_get_body (ctx->response, (guint8 **) & data, &size);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (_client, "Failed to get body of response...");
+ return;
+ }
+
+ GST_INFO_OBJECT (_client, "Response body is %d", size);
+ if (size > 0) {
+ if (!priv->m3_done) {
+ GstWFDResult wfd_res;
+ GstWFDMessage *msg = NULL;
+ /* Parse M3 response from sink */
+ wfd_res = gst_wfd_message_new (&msg);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client, "Failed to create wfd message...");
+ goto error;
+ }
+
+ wfd_res = gst_wfd_message_init (msg);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client, "Failed to init wfd message...");
+ goto error;
+ }
+
+ wfd_res = gst_wfd_message_parse_buffer (data, size, msg);
+
+ GST_DEBUG_OBJECT (client, "M3 response server side message body: %s",
+ gst_wfd_message_as_text (msg));
+
+ /* Get the audio formats supported by WFDSink */
+ if (msg->audio_codecs) {
+ wfd_res =
+ gst_wfd_message_get_supported_audio_format (msg, &priv->caCodec,
+ &priv->cFreq, &priv->cChanels, &priv->cBitwidth, &priv->caLatency);
+ if (wfd_res != GST_WFD_OK) {
+ GST_WARNING_OBJECT (client,
+ "Failed to get wfd support audio formats...");
+ goto error;
+ }
+ }
+
+ /* Get the Video formats supported by WFDSink */
+ wfd_res =
+ gst_wfd_message_get_supported_video_format (msg, &priv->cvCodec,
+ &priv->cNative, &priv->cNativeResolution,
+ (guint64 *) & priv->cCEAResolution,
+ (guint64 *) & priv->cVESAResolution,
+ (guint64 *) & priv->cHHResolution, &priv->cProfile, &priv->cLevel,
+ &priv->cvLatency, &priv->cMaxHeight, &priv->cMaxWidth,
+ &priv->cmin_slice_size, &priv->cslice_enc_params,
+ &priv->cframe_rate_control);
+ if (wfd_res != GST_WFD_OK) {
+ GST_WARNING_OBJECT (client,
+ "Failed to get wfd supported video formats...");
+ goto error;
+ }
+
+ if (msg->client_rtp_ports) {
+ /* Get the RTP ports preferred by WFDSink */
+ wfd_res =
+ gst_wfd_message_get_prefered_rtp_ports (msg, &priv->ctrans,
+ &priv->cprofile, &priv->clowertrans, &priv->crtp_port0,
+ &priv->crtp_port1);
+ if (wfd_res != GST_WFD_OK) {
+ GST_WARNING_OBJECT (client,
+ "Failed to get wfd prefered RTP ports...");
+ goto error;
+ }
+ }
+
+ if (msg->display_edid) {
+ guint32 edid_block_count = 0;
+ gchar *edid_payload = NULL;
+ priv->edid_supported = FALSE;
+ /* Get the display edid preferred by WFDSink */
+ GST_DEBUG_OBJECT (client, "Going to gst_wfd_message_get_display_edid");
+ wfd_res =
+ gst_wfd_message_get_display_edid (msg, &priv->edid_supported,
+ &edid_block_count, &edid_payload);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client, "Failed to get wfd display edid...");
+ goto error;
+ }
+ GST_DEBUG_OBJECT (client, " edid supported: %d edid_block_count: %d",
+ priv->edid_supported, edid_block_count);
+ if (priv->edid_supported) {
+ priv->edid_hres = 0;
+ priv->edid_vres = 0;
+ priv->edid_hres =
+ (guint32) (((edid_payload[54 + 4] >> 4) << 8) | edid_payload[54 +
+ 2]);
+ priv->edid_vres =
+ (guint32) (((edid_payload[54 + 7] >> 4) << 8) | edid_payload[54 +
+ 5]);
+ GST_DEBUG_OBJECT (client, " edid supported Hres: %d Wres: %d",
+ priv->edid_hres, priv->edid_vres);
+ if ((priv->edid_hres < 640) || (priv->edid_vres < 480)
+ || (priv->edid_hres > 1920) || (priv->edid_vres > 1080)) {
+ priv->edid_hres = 0;
+ priv->edid_vres = 0;
+ priv->edid_supported = FALSE;
+ GST_WARNING_OBJECT (client, " edid invalid resolutions");
+ }
+ }
+ }
+
+ if (msg->content_protection) {
+#if 0
+ /*Get the hdcp version and tcp port by WFDSink */
+ wfd_res =
+ gst_wfd_message_get_contentprotection_type (msg,
+ &priv->hdcp_version, &priv->hdcp_tcpport);
+ GST_DEBUG ("hdcp version =%d, tcp port = %d", priv->hdcp_version,
+ priv->hdcp_tcpport);
+ if (priv->hdcp_version > 0 && priv->hdcp_tcpport > 0)
+ priv->protection_enabled = TRUE;
+
+ if (wfd_res != GST_WFD_OK) {
+ GST_WARNING_OBJECT (client,
+ "Failed to get wfd content protection...");
+ goto error;
+ }
+#else
+ GST_WARNING_OBJECT (client, "Don't use content protection");
+#endif
+ }
+
+ g_signal_emit (_client,
+ gst_rtsp_client_wfd_signals[SIGNAL_WFD_GET_PARAMETER_REQUEST], 0,
+ ctx);
+ } else {
+ /* TODO-WFD: Handle another GET_PARAMETER response with body */
+ }
+ } else if (size == 0) {
+ if (!priv->m1_done) {
+ GST_INFO_OBJECT (_client, "M1 response is done");
+ priv->m1_done = TRUE;
+ } else if (!priv->m4_done) {
+ GST_INFO_OBJECT (_client, "M4 response is done");
+ priv->m4_done = TRUE;
+
+ gst_rtsp_wfd_client_trigger_request (_client, WFD_TRIGGER_SETUP);
+ } else {
+ g_mutex_lock(&priv->keep_alive_lock);
+ if (priv->keep_alive_flag == FALSE) {
+ GST_INFO_OBJECT (_client, "M16 response is done");
+ priv->keep_alive_flag = TRUE;
+ }
+ g_mutex_unlock(&priv->keep_alive_lock);
+ }
+ }
+
+ return;
+
+error:
+ return;
+}
+
+static gboolean
+handle_wfd_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPMethod options;
+ gchar *tmp = NULL;
+ gchar *str = NULL;
+ gchar *user_agent = NULL;
+
+ GstRTSPWFDClient *_client = GST_RTSP_WFD_CLIENT (client);
+
+ options = GST_RTSP_OPTIONS |
+ GST_RTSP_PAUSE |
+ GST_RTSP_PLAY |
+ GST_RTSP_SETUP |
+ GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
+
+ str = gst_rtsp_options_as_text (options);
+
+ /*append WFD specific method */
+ tmp = g_strdup (", org.wfa.wfd1.0");
+ g_strlcat (str, tmp, strlen (tmp) + strlen (str) + 1);
+
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
+ g_free (str);
+ str = NULL;
+
+ res =
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_USER_AGENT,
+ &user_agent, 0);
+ if (res == GST_RTSP_OK) {
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_USER_AGENT,
+ user_agent);
+ } else {
+ return FALSE;
+ }
+
+ res = gst_rtsp_client_send_message (client, NULL, ctx->response);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "gst_rtsp_client_send_message failed : %d", res);
+ return FALSE;
+ }
+
+ GST_DEBUG_OBJECT (client, "Sent M2 response...");
+
+ g_signal_emit (_client,
+ gst_rtsp_client_wfd_signals[SIGNAL_WFD_OPTIONS_REQUEST], 0, ctx);
+
+ return TRUE;
+}
+
+static gboolean
+handle_wfd_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ guint8 *data = NULL;
+ guint size = 0;
+
+ GstRTSPWFDClient *_client = GST_RTSP_WFD_CLIENT (client);
+
+ /* parsing the GET_PARAMTER request */
+ res = gst_rtsp_message_get_body (ctx->request, (guint8 **) & data, &size);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (_client, "Failed to get body of request...");
+ return FALSE;
+ }
+
+ if (size == 0) {
+ send_generic_wfd_response (_client, GST_RTSP_STS_OK, ctx);
+ } else {
+ /* TODO-WFD: Handle other GET_PARAMETER request from sink */
+ }
+
+ return TRUE;
+}
+
+static gboolean
+handle_wfd_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ guint8 *data = NULL;
+ guint size = 0;
+
+ GstRTSPWFDClient *_client = GST_RTSP_WFD_CLIENT (client);
+
+ res = gst_rtsp_message_get_body (ctx->request, &data, &size);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ if (size == 0) {
+ /* no body, keep-alive request */
+ send_generic_wfd_response (_client, GST_RTSP_STS_OK, ctx);
+ } else {
+ if (data != NULL) {
+ GST_INFO_OBJECT (_client, "SET_PARAMETER Request : %s(%d)", data, size);
+ if (g_strcmp0 ((const gchar *) data, "wfd_idr_request"))
+ send_generic_wfd_response (_client, GST_RTSP_STS_OK, ctx);
+#if 0
+ else
+ /* TODO-WFD : Handle other set param request */
+ send_generic_wfd_response (_client, GST_RTSP_STS_OK, ctx);
+#endif
+ } else {
+ goto bad_request;
+ }
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+bad_request:
+ {
+ GST_ERROR ("_client %p: bad request", _client);
+ send_generic_wfd_response (_client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+}
+
+#if 0
+static gboolean
+gst_rtsp_wfd_client_parse_methods (GstRTSPWFDClient * client,
+ GstRTSPMessage * response)
+{
+ GstRTSPHeaderField field;
+ gchar *respoptions;
+ gchar **options;
+ gint indx = 0;
+ gint i;
+ gboolean found_wfd_method = FALSE;
+
+ /* reset supported methods */
+ client->supported_methods = 0;
+
+ /* Try Allow Header first */
+ field = GST_RTSP_HDR_ALLOW;
+ while (TRUE) {
+ respoptions = NULL;
+ gst_rtsp_message_get_header (response, field, &respoptions, indx);
+ if (indx == 0 && !respoptions) {
+ /* if no Allow header was found then try the Public header... */
+ field = GST_RTSP_HDR_PUBLIC;
+ gst_rtsp_message_get_header (response, field, &respoptions, indx);
+ }
+ if (!respoptions)
+ break;
+
+ /* If we get here, the server gave a list of supported methods, parse
+ * them here. The string is like:
+ *
+ * OPTIONS, PLAY, SETUP, ...
+ */
+ options = g_strsplit (respoptions, ",", 0);
+
+ for (i = 0; options[i]; i++) {
+ gchar *stripped;
+ gint method;
+
+ stripped = g_strstrip (options[i]);
+ method = gst_rtsp_find_method (stripped);
+
+ if (!g_ascii_strcasecmp ("org.wfa.wfd1.0", stripped))
+ found_wfd_method = TRUE;
+
+ /* keep bitfield of supported methods */
+ if (method != GST_RTSP_INVALID)
+ client->supported_methods |= method;
+ }
+ g_strfreev (options);
+
+ indx++;
+ }
+
+ if (!found_wfd_method) {
+ GST_ERROR_OBJECT (client,
+ "WFD client is not supporting WFD mandatory message : org.wfa.wfd1.0...");
+ goto no_required_methods;
+ }
+
+ /* Checking mandatory method */
+ if (!(client->supported_methods & GST_RTSP_SET_PARAMETER)) {
+ GST_ERROR_OBJECT (client,
+ "WFD client is not supporting WFD mandatory message : SET_PARAMETER...");
+ goto no_required_methods;
+ }
+
+ /* Checking mandatory method */
+ if (!(client->supported_methods & GST_RTSP_GET_PARAMETER)) {
+ GST_ERROR_OBJECT (client,
+ "WFD client is not supporting WFD mandatory message : GET_PARAMETER...");
+ goto no_required_methods;
+ }
+
+ if (!(client->supported_methods & GST_RTSP_OPTIONS)) {
+ GST_INFO_OBJECT (client, "assuming OPTIONS is supported by client...");
+ client->supported_methods |= GST_RTSP_OPTIONS;
+ }
+
+ return TRUE;
+
+/* ERRORS */
+no_required_methods:
+ {
+ GST_ELEMENT_ERROR (client, RESOURCE, OPEN_READ, (NULL),
+ ("WFD Client does not support mandatory methods."));
+ return FALSE;
+ }
+}
+#endif
+
+typedef enum
+{
+ M1_REQ_MSG,
+ M1_RES_MSG,
+ M2_REQ_MSG,
+ M2_RES_MSG,
+ M3_REQ_MSG,
+ M3_RES_MSG,
+ M4_REQ_MSG,
+ M4_RES_MSG,
+ M5_REQ_MSG,
+ TEARDOWN_TRIGGER,
+ PLAY_TRIGGER,
+ PAUSE_TRIGGER,
+} GstWFDMessageType;
+
+static gboolean
+_set_negotiated_audio_codec (GstRTSPWFDClient *client,
+ guint audio_codec)
+{
+ GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+
+ GstRTSPMediaFactory *factory = NULL;
+ GstRTSPMountPoints *mount_points = NULL;
+ gchar *path = NULL;
+ gint matched = 0;
+ gboolean ret = TRUE;
+
+ if (!(mount_points = gst_rtsp_client_get_mount_points (parent_client))) {
+ ret = FALSE;
+ GST_ERROR_OBJECT (client, "Failed to set negotiated audio codec: no mount points...");
+ goto no_mount_points;
+ }
+
+ path = g_strdup(WFD_MOUNT_POINT);
+ if (!path) {
+ ret = FALSE;
+ GST_ERROR_OBJECT (client, "Failed to set negotiated audio codec: no path...");
+ goto no_path;
+ }
+
+ if (!(factory = gst_rtsp_mount_points_match (mount_points,
+ path, &matched))) {
+ GST_ERROR_OBJECT (client, "Failed to set negotiated audio codec: no factory...");
+ ret = FALSE;
+ goto no_factory;
+ }
+
+ gst_rtsp_media_factory_wfd_set_audio_codec (factory,
+ audio_codec);
+ ret = TRUE;
+
+ g_object_unref(factory);
+
+no_factory:
+ g_free(path);
+no_path:
+ g_object_unref(mount_points);
+no_mount_points:
+ return ret;
+}
+
+static gboolean
+_set_negotiated_resolution(GstRTSPWFDClient *client,
+ guint32 width, guint32 height)
+{
+ GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+
+ GstRTSPMediaFactory *factory = NULL;
+ GstRTSPMountPoints *mount_points = NULL;
+ gchar *path = NULL;
+ gint matched = 0;
+ gboolean ret = TRUE;
+
+ if (!(mount_points = gst_rtsp_client_get_mount_points (parent_client))) {
+ ret = FALSE;
+ GST_ERROR_OBJECT (client, "Failed to set negotiated resolution: no mount points...");
+ goto no_mount_points;
+ }
+
+ path = g_strdup(WFD_MOUNT_POINT);
+ if (!path) {
+ ret = FALSE;
+ GST_ERROR_OBJECT (client, "Failed to set negotiated resolution: no path...");
+ goto no_path;
+ }
+
+ if (!(factory = gst_rtsp_mount_points_match (mount_points,
+ path, &matched))) {
+ GST_ERROR_OBJECT (client, "Failed to set negotiated resolution: no factory...");
+ ret = FALSE;
+ goto no_factory;
+ }
+
+ gst_rtsp_media_factory_wfd_set_negotiated_resolution(factory,
+ width, height);
+ ret = TRUE;
+
+ g_object_unref(factory);
+
+no_factory:
+ g_free(path);
+no_path:
+ g_object_unref(mount_points);
+no_mount_points:
+ return ret;
+}
+
+static void
+_set_wfd_message_body (GstRTSPWFDClient * client, GstWFDMessageType msg_type,
+ gchar ** data, guint * len)
+{
+ GString *buf = NULL;
+ GstWFDMessage *msg = NULL;
+ GstWFDResult wfd_res = GST_WFD_EINVAL;
+ GstRTSPWFDClientPrivate *priv = NULL;
+ priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ g_return_if_fail (priv != NULL);
+
+ buf = g_string_new ("");
+ g_return_if_fail (buf != NULL);
+
+ if (msg_type == M3_REQ_MSG) {
+ /* create M3 request to be sent */
+ wfd_res = gst_wfd_message_new (&msg);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client, "Failed to create wfd message...");
+ goto error;
+ }
+
+ wfd_res = gst_wfd_message_init (msg);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client, "Failed to init wfd message...");
+ goto error;
+ }
+
+ /* set the supported audio formats by the WFD server */
+ wfd_res =
+ gst_wfd_message_set_supported_audio_format (msg, GST_WFD_AUDIO_UNKNOWN,
+ GST_WFD_FREQ_UNKNOWN, GST_WFD_CHANNEL_UNKNOWN, 0, 0);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client,
+ "Failed to set supported audio formats on wfd message...");
+ goto error;
+ }
+
+ /* set the supported Video formats by the WFD server */
+ wfd_res =
+ gst_wfd_message_set_supported_video_format (msg, GST_WFD_VIDEO_UNKNOWN,
+ GST_WFD_VIDEO_CEA_RESOLUTION, GST_WFD_CEA_UNKNOWN, GST_WFD_CEA_UNKNOWN,
+ GST_WFD_VESA_UNKNOWN, GST_WFD_HH_UNKNOWN, GST_WFD_H264_UNKNOWN_PROFILE,
+ GST_WFD_H264_LEVEL_UNKNOWN, 0, 0, 0, 0, 0, 0);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client,
+ "Failed to set supported video formats on wfd message...");
+ goto error;
+ }
+
+ wfd_res = gst_wfd_message_set_display_edid (msg, 0, 0, NULL);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client,
+ "Failed to set display edid type on wfd message...");
+ goto error;
+ }
+
+ if (priv->protection_enabled) {
+ wfd_res =
+ gst_wfd_message_set_contentprotection_type (msg, GST_WFD_HDCP_NONE,
+ 0);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client,
+ "Failed to set supported content protection type on wfd message...");
+ goto error;
+ }
+ }
+
+ /* set the preffered RTP ports for the WFD server */
+ wfd_res =
+ gst_wfd_messge_set_prefered_rtp_ports (msg, GST_WFD_RTSP_TRANS_UNKNOWN,
+ GST_WFD_RTSP_PROFILE_UNKNOWN, GST_WFD_RTSP_LOWER_TRANS_UNKNOWN, 0, 0);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client,
+ "Failed to set supported video formats on wfd message...");
+ goto error;
+ }
+
+ *data = gst_wfd_message_param_names_as_text (msg);
+ if (*data == NULL) {
+ GST_ERROR_OBJECT (client, "Failed to get wfd message as text...");
+ goto error;
+ } else {
+ *len = strlen (*data);
+ }
+ } else if (msg_type == M4_REQ_MSG) {
+ GstRTSPUrl *url = NULL;
+
+ GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+ GstRTSPConnection *connection =
+ gst_rtsp_client_get_connection (parent_client);
+
+ /* Parameters for the preffered audio formats */
+ GstWFDAudioFormats taudiocodec = GST_WFD_AUDIO_UNKNOWN;
+ GstWFDAudioFreq taudiofreq = GST_WFD_FREQ_UNKNOWN;
+ GstWFDAudioChannels taudiochannels = GST_WFD_CHANNEL_UNKNOWN;
+
+ /* Parameters for the preffered video formats */
+ GstWFDVideoCEAResolution tcCEAResolution = GST_WFD_CEA_UNKNOWN;
+ GstWFDVideoVESAResolution tcVESAResolution = GST_WFD_VESA_UNKNOWN;
+ GstWFDVideoHHResolution tcHHResolution = GST_WFD_HH_UNKNOWN;
+ GstWFDVideoH264Profile tcProfile;
+ GstWFDVideoH264Level tcLevel;
+ guint64 resolution_supported = 0;
+
+ url = gst_rtsp_connection_get_url (connection);
+ if (url == NULL) {
+ GST_ERROR_OBJECT (client, "Failed to get connection URL");
+ return;
+ }
+
+ /* Logic to negotiate with information of M3 response */
+ /* create M4 request to be sent */
+ wfd_res = gst_wfd_message_new (&msg);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client, "Failed to create wfd message...");
+ goto error;
+ }
+
+ wfd_res = gst_wfd_message_init (msg);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client, "Failed to init wfd message...");
+ goto error;
+ }
+
+ g_string_append_printf (buf, "rtsp://");
+
+ if (priv->host_address) {
+ g_string_append (buf, priv->host_address);
+ } else {
+ GST_ERROR_OBJECT (client, "Failed to get host address");
+ if (buf) g_string_free (buf, TRUE);
+ goto error;
+ }
+
+ g_string_append_printf (buf, "/wfd1.0/streamid=0");
+ wfd_res =
+ gst_wfd_message_set_presentation_url (msg, g_string_free (buf, FALSE),
+ NULL);
+
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client, "Failed to set presentation url");
+ goto error;
+ }
+
+ taudiocodec = wfd_get_prefered_audio_codec (priv->audio_codec, priv->caCodec);
+ priv->caCodec = taudiocodec;
+ if (!_set_negotiated_audio_codec(client, priv->caCodec)) {
+ GST_ERROR_OBJECT (client, "Failed to set negotiated "
+ "audio codec to media factory...");
+ }
+
+ if (priv->cFreq & GST_WFD_FREQ_48000)
+ taudiofreq = GST_WFD_FREQ_48000;
+ else if (priv->cFreq & GST_WFD_FREQ_44100)
+ taudiofreq = GST_WFD_FREQ_44100;
+ priv->cFreq = taudiofreq;
+
+ /* TODO-WFD: Currently only 2 channels is present */
+ if (priv->cChanels & GST_WFD_CHANNEL_8)
+ taudiochannels = GST_WFD_CHANNEL_2;
+ else if (priv->cChanels & GST_WFD_CHANNEL_6)
+ taudiochannels = GST_WFD_CHANNEL_2;
+ else if (priv->cChanels & GST_WFD_CHANNEL_4)
+ taudiochannels = GST_WFD_CHANNEL_2;
+ else if (priv->cChanels & GST_WFD_CHANNEL_2)
+ taudiochannels = GST_WFD_CHANNEL_2;
+ priv->cChanels = taudiochannels;
+
+ wfd_res =
+ gst_wfd_message_set_prefered_audio_format (msg, taudiocodec, taudiofreq,
+ taudiochannels, priv->cBitwidth, priv->caLatency);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (priv, "Failed to set preffered audio formats...");
+ goto error;
+ }
+
+ /* Set the preffered video formats */
+ priv->cvCodec = GST_WFD_VIDEO_H264;
+ priv->cProfile = tcProfile = GST_WFD_H264_BASE_PROFILE;
+ priv->cLevel = tcLevel = GST_WFD_H264_LEVEL_3_1;
+
+ resolution_supported = priv->video_resolution_supported;
+
+ /* TODO-WFD: Need to verify this logic
+ if(priv->edid_supported) {
+ if (priv->edid_hres < 1920) resolution_supported = resolution_supported & 0x8C7F;
+ if (priv->edid_hres < 1280) resolution_supported = resolution_supported & 0x1F;
+ if (priv->edid_hres < 720) resolution_supported = resolution_supported & 0x01;
+ }
+ */
+
+ if (priv->video_native_resolution == GST_WFD_VIDEO_CEA_RESOLUTION) {
+ tcCEAResolution =
+ wfd_get_prefered_resolution (resolution_supported,
+ priv->cCEAResolution, priv->video_native_resolution, &priv->cMaxWidth,
+ &priv->cMaxHeight, &priv->cFramerate, &priv->cInterleaved);
+ GST_DEBUG
+ ("wfd negotiated resolution: %08x, width: %d, height: %d, framerate: %d, interleaved: %d",
+ tcCEAResolution, priv->cMaxWidth, priv->cMaxHeight, priv->cFramerate,
+ priv->cInterleaved);
+ } else if (priv->video_native_resolution == GST_WFD_VIDEO_VESA_RESOLUTION) {
+ tcVESAResolution =
+ wfd_get_prefered_resolution (resolution_supported,
+ priv->cVESAResolution, priv->video_native_resolution,
+ &priv->cMaxWidth, &priv->cMaxHeight, &priv->cFramerate,
+ &priv->cInterleaved);
+ GST_DEBUG
+ ("wfd negotiated resolution: %08x, width: %d, height: %d, framerate: %d, interleaved: %d",
+ tcVESAResolution, priv->cMaxWidth, priv->cMaxHeight, priv->cFramerate,
+ priv->cInterleaved);
+ } else if (priv->video_native_resolution == GST_WFD_VIDEO_HH_RESOLUTION) {
+ tcHHResolution =
+ wfd_get_prefered_resolution (resolution_supported,
+ priv->cHHResolution, priv->video_native_resolution, &priv->cMaxWidth,
+ &priv->cMaxHeight, &priv->cFramerate, &priv->cInterleaved);
+ GST_DEBUG
+ ("wfd negotiated resolution: %08x, width: %d, height: %d, framerate: %d, interleaved: %d",
+ tcHHResolution, priv->cMaxWidth, priv->cMaxHeight, priv->cFramerate,
+ priv->cInterleaved);
+ }
+
+ if (!_set_negotiated_resolution(client, priv->cMaxWidth,
+ priv->cMaxHeight)) {
+ GST_ERROR_OBJECT (client, "Failed to set negotiated "
+ "resolution to media factory...");
+ }
+
+ wfd_res =
+ gst_wfd_message_set_prefered_video_format (msg, priv->cvCodec,
+ priv->video_native_resolution, GST_WFD_CEA_UNKNOWN, tcCEAResolution,
+ tcVESAResolution, tcHHResolution, tcProfile, tcLevel, priv->cvLatency,
+ priv->cMaxWidth, priv->cMaxHeight, priv->cmin_slice_size,
+ priv->cslice_enc_params, priv->cframe_rate_control);
+
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client, "Failed to set preffered video formats...");
+ goto error;
+ }
+
+ /* set the preffered RTP ports for the WFD server */
+ wfd_res =
+ gst_wfd_messge_set_prefered_rtp_ports (msg, GST_WFD_RTSP_TRANS_RTP,
+ GST_WFD_RTSP_PROFILE_AVP, GST_WFD_RTSP_LOWER_TRANS_UDP, priv->crtp_port0, priv->crtp_port1);
+ if (wfd_res != GST_WFD_OK) {
+ GST_ERROR_OBJECT (client,
+ "Failed to set supported video formats on wfd message...");
+ goto error;
+ }
+
+ *data = gst_wfd_message_as_text (msg);
+ if (*data == NULL) {
+ GST_ERROR_OBJECT (client, "Failed to get wfd message as text...");
+ goto error;
+ } else {
+ *len = strlen (*data);
+ }
+ } else if (msg_type == M5_REQ_MSG) {
+ g_string_append (buf, "wfd_trigger_method: SETUP");
+ g_string_append (buf, "\r\n");
+ *len = buf->len;
+ *data = g_string_free (buf, FALSE);
+ } else if (msg_type == TEARDOWN_TRIGGER) {
+ g_string_append (buf, "wfd_trigger_method: TEARDOWN");
+ g_string_append (buf, "\r\n");
+ *len = buf->len;
+ *data = g_string_free (buf, FALSE);
+ } else if (msg_type == PLAY_TRIGGER) {
+ g_string_append (buf, "wfd_trigger_method: PLAY");
+ g_string_append (buf, "\r\n");
+ *len = buf->len;
+ *data = g_string_free (buf, FALSE);
+ } else if (msg_type == PAUSE_TRIGGER) {
+ g_string_append (buf, "wfd_trigger_method: PAUSE");
+ g_string_append (buf, "\r\n");
+ *len = buf->len;
+ *data = g_string_free (buf, FALSE);
+ } else {
+ return;
+ }
+
+ return;
+
+error:
+ *data = NULL;
+ *len = 0;
+
+ return;
+}
+
+/**
+* prepare_request:
+* @client: client object
+* @request : requst message to be prepared
+* @method : RTSP method of the request
+* @url : url need to be in the request
+* @message_type : WFD message type
+* @trigger_type : trigger method to be used for M5 mainly
+*
+* Prepares request based on @method & @message_type
+*
+* Returns: a #GstRTSPResult.
+*/
+GstRTSPResult
+prepare_request (GstRTSPWFDClient * client, GstRTSPMessage * request,
+ GstRTSPMethod method, gchar * url)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ gchar *str = NULL;
+
+ if (method == GST_RTSP_GET_PARAMETER || method == GST_RTSP_SET_PARAMETER) {
+ g_free (url);
+ url = g_strdup ("rtsp://localhost/wfd1.0");
+ }
+
+ GST_DEBUG_OBJECT (client, "Preparing request: %d", method);
+
+ /* initialize the request */
+ res = gst_rtsp_message_init_request (request, method, url);
+ if (res < 0) {
+ GST_ERROR ("init request failed");
+ return res;
+ }
+
+ switch (method) {
+ /* Prepare OPTIONS request to send */
+ case GST_RTSP_OPTIONS:{
+ /* add wfd specific require filed "org.wfa.wfd1.0" */
+ str = g_strdup ("org.wfa.wfd1.0");
+ res = gst_rtsp_message_add_header (request, GST_RTSP_HDR_REQUIRE, str);
+ if (res < 0) {
+ GST_ERROR ("Failed to add header");
+ g_free (str);
+ return res;
+ }
+
+ g_free (str);
+ break;
+ }
+
+ /* Prepare GET_PARAMETER request */
+ case GST_RTSP_GET_PARAMETER:{
+ gchar *msg = NULL;
+ guint msglen = 0;
+ GString *msglength;
+
+ /* add content type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "text/parameters");
+ if (res < 0) {
+ GST_ERROR ("Failed to add header");
+ return res;
+ }
+
+ _set_wfd_message_body (client, M3_REQ_MSG, &msg, &msglen);
+ msglength = g_string_new ("");
+ g_string_append_printf (msglength, "%d", msglen);
+ GST_DEBUG ("M3 server side message body: %s", msg);
+
+ /* add content-length type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+ g_string_free (msglength, FALSE));
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ g_free (msg);
+ break;
+ }
+
+ /* Prepare SET_PARAMETER request */
+ case GST_RTSP_SET_PARAMETER:{
+ gchar *msg = NULL;
+ guint msglen = 0;
+ GString *msglength;
+
+ /* add content type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "text/parameters");
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp request...");
+ goto error;
+ }
+
+ _set_wfd_message_body (client, M4_REQ_MSG, &msg, &msglen);
+ msglength = g_string_new ("");
+ g_string_append_printf (msglength, "%d", msglen);
+ GST_DEBUG ("M4 server side message body: %s", msg);
+
+ /* add content-length type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+ g_string_free (msglength, FALSE));
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ g_free (msg);
+ break;
+ }
+
+ default:{
+ }
+ }
+
+ return res;
+
+error:
+ return GST_RTSP_ERROR;
+}
+
+GstRTSPResult
+prepare_trigger_request (GstRTSPWFDClient * client, GstRTSPMessage * request,
+ GstWFDTriggerType trigger_type, gchar * url)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+
+ /* initialize the request */
+ res = gst_rtsp_message_init_request (request, GST_RTSP_SET_PARAMETER, url);
+ if (res < 0) {
+ GST_ERROR ("init request failed");
+ return res;
+ }
+
+ switch (trigger_type) {
+ case WFD_TRIGGER_SETUP:{
+ gchar *msg;
+ guint msglen = 0;
+ GString *msglength;
+
+ /* add content type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "text/parameters");
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp request...");
+ goto error;
+ }
+
+ _set_wfd_message_body (client, M5_REQ_MSG, &msg, &msglen);
+ msglength = g_string_new ("");
+ g_string_append_printf (msglength, "%d", msglen);
+ GST_DEBUG ("M5 server side message body: %s", msg);
+
+ /* add content-length type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+ g_string_free (msglength, FALSE));
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ g_free (msg);
+ break;
+ }
+ case WFD_TRIGGER_TEARDOWN:{
+ gchar *msg;
+ guint msglen = 0;
+ GString *msglength;
+
+ /* add content type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "text/parameters");
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp request...");
+ goto error;
+ }
+
+ _set_wfd_message_body (client, TEARDOWN_TRIGGER, &msg, &msglen);
+ msglength = g_string_new ("");
+ g_string_append_printf (msglength, "%d", msglen);
+ GST_DEBUG ("Trigger TEARDOWN server side message body: %s", msg);
+
+ /* add content-length type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+ g_string_free (msglength, FALSE));
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ g_free (msg);
+ break;
+ }
+ case WFD_TRIGGER_PLAY:{
+ gchar *msg;
+ guint msglen = 0;
+ GString *msglength;
+
+ /* add content type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "text/parameters");
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp request...");
+ goto error;
+ }
+
+ _set_wfd_message_body (client, PLAY_TRIGGER, &msg, &msglen);
+ msglength = g_string_new ("");
+ g_string_append_printf (msglength, "%d", msglen);
+ GST_DEBUG ("Trigger PLAY server side message body: %s", msg);
+
+ /* add content-length type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+ g_string_free (msglength, FALSE));
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ g_free (msg);
+ break;
+ }
+ case WFD_TRIGGER_PAUSE:{
+ gchar *msg;
+ guint msglen = 0;
+ GString *msglength;
+
+ /* add content type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "text/parameters");
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp request...");
+ goto error;
+ }
+
+ _set_wfd_message_body (client, PAUSE_TRIGGER, &msg, &msglen);
+ msglength = g_string_new ("");
+ g_string_append_printf (msglength, "%d", msglen);
+ GST_DEBUG ("Trigger PAUSE server side message body: %s", msg);
+
+ /* add content-length type */
+ res =
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+ g_string_free (msglength, FALSE));
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+ goto error;
+ }
+
+ g_free (msg);
+ break;
+ }
+ /* TODO-WFD: implement to handle other trigger type */
+ default:{
+ }
+ }
+
+ return res;
+
+error:
+ return res;
+}
+
+
+void
+send_request (GstRTSPWFDClient * client, GstRTSPSession * session,
+ GstRTSPMessage * request)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+
+ /* remove any previous header */
+ gst_rtsp_message_remove_header (request, GST_RTSP_HDR_SESSION, -1);
+
+ /* add the new session header for new session ids */
+ if (session) {
+ guint timeout;
+ const gchar *sessionid = NULL;
+ gchar *str;
+
+ sessionid = gst_rtsp_session_get_sessionid (session);
+ GST_INFO_OBJECT (client, "Session id : %s", sessionid);
+
+ timeout = gst_rtsp_session_get_timeout (session);
+ if (timeout != DEFAULT_WFD_TIMEOUT)
+ str = g_strdup_printf ("%s; timeout=%d", sessionid, timeout);
+ else
+ str = g_strdup (sessionid);
+
+ gst_rtsp_message_take_header (request, GST_RTSP_HDR_SESSION, str);
+ }
+#if 0
+ if (gst_debug_category_get_threshold (rtsp_wfd_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (request);
+ }
+#endif
+ res = gst_rtsp_client_send_message (parent_client, session, request);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "gst_rtsp_client_send_message failed : %d", res);
+ }
+
+ gst_rtsp_message_unset (request);
+}
+
+/**
+* prepare_response:
+* @client: client object
+* @request : requst message received
+* @response : response to be prepare based on request
+* @method : RTSP method
+*
+* prepare response to the request based on @method & @message_type
+*
+* Returns: a #GstRTSPResult.
+*/
+GstRTSPResult
+prepare_response (GstRTSPWFDClient * client, GstRTSPMessage * request,
+ GstRTSPMessage * response, GstRTSPMethod method)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+
+ switch (method) {
+ /* prepare OPTIONS response */
+ case GST_RTSP_OPTIONS:{
+ GstRTSPMethod options;
+ gchar *tmp = NULL;
+ gchar *str = NULL;
+ gchar *user_agent = NULL;
+
+ options = GST_RTSP_OPTIONS |
+ GST_RTSP_PAUSE |
+ GST_RTSP_PLAY |
+ GST_RTSP_SETUP |
+ GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
+
+ str = gst_rtsp_options_as_text (options);
+
+ /*append WFD specific method */
+ tmp = g_strdup (", org.wfa.wfd1.0");
+ g_strlcat (str, tmp, strlen (tmp) + strlen (str) + 1);
+
+ gst_rtsp_message_init_response (response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
+
+ gst_rtsp_message_add_header (response, GST_RTSP_HDR_PUBLIC, str);
+ g_free (str);
+ str = NULL;
+ res =
+ gst_rtsp_message_get_header (request, GST_RTSP_HDR_USER_AGENT,
+ &user_agent, 0);
+ if (res == GST_RTSP_OK) {
+ gst_rtsp_message_add_header (response, GST_RTSP_HDR_USER_AGENT,
+ user_agent);
+ } else
+ res = GST_RTSP_OK;
+ break;
+ }
+ default:
+ GST_ERROR_OBJECT (client, "Unhandled method...");
+ return GST_RTSP_EINVAL;
+ break;
+ }
+
+ return res;
+}
+
+static void
+send_generic_wfd_response (GstRTSPWFDClient * client, GstRTSPStatusCode code,
+ GstRTSPContext * ctx)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ res = gst_rtsp_client_send_message (parent_client, NULL, ctx->response);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (client, "gst_rtsp_client_send_message failed : %d", res);
+ }
+}
+
+
+static GstRTSPResult
+handle_M1_message (GstRTSPWFDClient * client)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPMessage request = { 0 };
+
+ res = prepare_request (client, &request, GST_RTSP_OPTIONS, (gchar *) "*");
+ if (GST_RTSP_OK != res) {
+ GST_ERROR_OBJECT (client, "Failed to prepare M1 request....\n");
+ return res;
+ }
+
+ GST_DEBUG_OBJECT (client, "Sending M1 request.. (OPTIONS request)");
+
+ send_request (client, NULL, &request);
+
+ return res;
+}
+
+/**
+* handle_M3_message:
+* @client: client object
+*
+* Handles M3 WFD message.
+* This API will send M3 message (GET_PARAMETER) to WFDSink to query supported formats by the WFDSink.
+* After getting supported formats info, this API will set those values on WFDConfigMessage obj
+*
+* Returns: a #GstRTSPResult.
+*/
+static GstRTSPResult
+handle_M3_message (GstRTSPWFDClient * client)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPMessage request = { 0 };
+ GstRTSPUrl *url = NULL;
+ gchar *url_str = NULL;
+
+ GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+ GstRTSPConnection *connection =
+ gst_rtsp_client_get_connection (parent_client);
+
+ url = gst_rtsp_connection_get_url (connection);
+ if (url == NULL) {
+ GST_ERROR_OBJECT (client, "Failed to get connection URL");
+ res = GST_RTSP_ERROR;
+ goto error;
+ }
+
+ url_str = gst_rtsp_url_get_request_uri (url);
+ if (url_str == NULL) {
+ GST_ERROR_OBJECT (client, "Failed to get connection URL");
+ res = GST_RTSP_ERROR;
+ goto error;
+ }
+
+ res = prepare_request (client, &request, GST_RTSP_GET_PARAMETER, url_str);
+ if (GST_RTSP_OK != res) {
+ GST_ERROR_OBJECT (client, "Failed to prepare M3 request....\n");
+ goto error;
+ }
+
+ GST_DEBUG_OBJECT (client, "Sending GET_PARAMETER request message (M3)...");
+
+ send_request (client, NULL, &request);
+
+ return res;
+
+error:
+ return res;
+}
+
+static GstRTSPResult
+handle_M4_message (GstRTSPWFDClient * client)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPMessage request = { 0 };
+ GstRTSPUrl *url = NULL;
+ gchar *url_str = NULL;
+
+ GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+ GstRTSPConnection *connection =
+ gst_rtsp_client_get_connection (parent_client);
+
+ url = gst_rtsp_connection_get_url (connection);
+ if (url == NULL) {
+ GST_ERROR_OBJECT (client, "Failed to get connection URL");
+ res = GST_RTSP_ERROR;
+ goto error;
+ }
+
+ url_str = gst_rtsp_url_get_request_uri (url);
+ if (url_str == NULL) {
+ GST_ERROR_OBJECT (client, "Failed to get connection URL");
+ res = GST_RTSP_ERROR;
+ goto error;
+ }
+
+ res = prepare_request (client, &request, GST_RTSP_SET_PARAMETER, url_str);
+ if (GST_RTSP_OK != res) {
+ GST_ERROR_OBJECT (client, "Failed to prepare M4 request....\n");
+ goto error;
+ }
+
+ GST_DEBUG_OBJECT (client, "Sending SET_PARAMETER request message (M4)...");
+
+ send_request (client, NULL, &request);
+
+ return res;
+
+error:
+ return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_client_trigger_request (GstRTSPWFDClient * client,
+ GstWFDTriggerType type)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPMessage request = { 0 };
+ GstRTSPUrl *url = NULL;
+ gchar *url_str = NULL;
+
+ GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+ GstRTSPConnection *connection =
+ gst_rtsp_client_get_connection (parent_client);
+
+ url = gst_rtsp_connection_get_url (connection);
+ if (url == NULL) {
+ GST_ERROR_OBJECT (client, "Failed to get connection URL");
+ res = GST_RTSP_ERROR;
+ goto error;
+ }
+
+ url_str = gst_rtsp_url_get_request_uri (url);
+ if (url_str == NULL) {
+ GST_ERROR_OBJECT (client, "Failed to get connection URL");
+ res = GST_RTSP_ERROR;
+ goto error;
+ }
+
+ res = prepare_trigger_request (client, &request, type, url_str);
+ if (GST_RTSP_OK != res) {
+ GST_ERROR_OBJECT (client, "Failed to prepare M5 request....\n");
+ goto error;
+ }
+
+ GST_DEBUG_OBJECT (client, "Sending trigger request message...: %d", type);
+
+ send_request (client, NULL, &request);
+
+ return res;
+
+error:
+ return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_client_set_video_supported_resolution (GstRTSPWFDClient * client,
+ guint64 supported_reso)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ g_return_val_if_fail (priv != NULL, GST_RTSP_EINVAL);
+
+ priv->video_resolution_supported = supported_reso;
+ GST_DEBUG ("Resolution : %"G_GUINT64_FORMAT, supported_reso);
+
+ return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_client_set_video_native_resolution (GstRTSPWFDClient * client,
+ guint64 native_reso)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ g_return_val_if_fail (priv != NULL, GST_RTSP_EINVAL);
+
+ priv->video_native_resolution = native_reso;
+ GST_DEBUG ("Native Resolution : %"G_GUINT64_FORMAT, native_reso);
+
+ return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_client_set_audio_codec (GstRTSPWFDClient * client,
+ guint8 audio_codec)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ g_return_val_if_fail (priv != NULL, GST_RTSP_EINVAL);
+
+ priv->audio_codec = audio_codec;
+ GST_DEBUG ("Audio codec : %d", audio_codec);
+
+ return res;
+}
+
+static gboolean
+wfd_ckeck_keep_alive_response (gpointer userdata)
+{
+ GstRTSPWFDClient *client = (GstRTSPWFDClient *)userdata;
+ GstRTSPWFDClientPrivate *priv = NULL;
+ if (!client) {
+ return FALSE;
+ }
+
+ priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, GST_RTSP_EINVAL);
+
+ if (priv->keep_alive_flag) {
+ return FALSE;
+ }
+ else {
+ GST_INFO ("%p: source error notification", client);
+ // FIXME Do something here. Maybe emit some signal?
+ return FALSE;
+ }
+}
+
+/*Sending keep_alive (M16) message.
+ Without calling prepare_request function.*/
+static GstRTSPResult
+handle_M16_message (GstRTSPWFDClient * client)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPMessage request = { 0 };
+ gchar *url_str = NULL;
+
+ url_str = g_strdup("rtsp://localhost/wfd1.0");
+
+ res = gst_rtsp_message_init_request (&request, GST_RTSP_GET_PARAMETER, url_str);
+ if (res < 0) {
+ GST_ERROR ("init request failed");
+ return FALSE;
+ }
+
+ send_request (client, NULL, &request);
+ return GST_RTSP_OK;
+}
+
+/*CHecking whether source has got response of any request.
+ * If yes, keep alive message is sent otherwise error message
+ * will be displayed.*/
+static gboolean
+keep_alive_condition(gpointer userdata)
+{
+ GstRTSPWFDClient *client;
+ GstRTSPWFDClientPrivate *priv;
+ GstRTSPResult res;
+ client = (GstRTSPWFDClient *)userdata;
+ if (!client) {
+ return FALSE;
+ }
+ priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ g_return_val_if_fail (priv != NULL, FALSE);
+
+ g_mutex_lock(&priv->keep_alive_lock);
+ if(!priv->keep_alive_flag) {
+ g_timeout_add(5000, wfd_ckeck_keep_alive_response, client);
+ }
+ else {
+ GST_DEBUG_OBJECT (client, "have received last keep alive message response");
+ }
+
+ GST_DEBUG("sending keep alive message");
+ res = handle_M16_message(client);
+ if(res == GST_RTSP_OK) {
+ priv->keep_alive_flag = FALSE;
+ } else {
+ GST_ERROR_OBJECT (client, "Failed to send Keep Alive Message");
+ g_mutex_unlock(&priv->keep_alive_lock);
+ return FALSE;
+ }
+
+ g_mutex_unlock(&priv->keep_alive_lock);
+ return TRUE;
+}
+
+static
+void wfd_set_keep_alive_condition(GstRTSPWFDClient * client)
+{
+ g_timeout_add((DEFAULT_WFD_TIMEOUT-5)*1000, keep_alive_condition, client);
+}
+
+void
+gst_rtsp_wfd_client_set_host_address (GstRTSPWFDClient *client, const gchar * address)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+ g_return_if_fail (priv != NULL);
+
+ if (priv->host_address) {
+ g_free (priv->host_address);
+ }
+
+ priv->host_address = g_strdup (address);
+}
+
+guint
+gst_rtsp_wfd_client_get_audio_codec(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->caCodec;
+}
+
+guint
+gst_rtsp_wfd_client_get_audio_freq(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cFreq;
+}
+
+guint
+gst_rtsp_wfd_client_get_audio_channels(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cChanels;
+}
+
+guint
+gst_rtsp_wfd_client_get_audio_bit_width(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cBitwidth;
+}
+
+guint
+gst_rtsp_wfd_client_get_audio_latency(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->caLatency;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_codec(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cvCodec;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_native(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cNative;
+}
+
+guint64
+gst_rtsp_wfd_client_get_video_native_resolution(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cNativeResolution;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_cea_resolution(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cCEAResolution;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_vesa_resolution(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cVESAResolution;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_hh_resolution(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cHHResolution;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_profile(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cProfile;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_level(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cLevel;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_latency(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cvLatency;
+}
+
+guint32
+gst_rtsp_wfd_client_get_video_max_height(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cMaxHeight;
+}
+
+guint32
+gst_rtsp_wfd_client_get_video_max_width(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cMaxWidth;
+}
+
+guint32
+gst_rtsp_wfd_client_get_video_framerate(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cFramerate;
+}
+
+guint32
+gst_rtsp_wfd_client_get_video_min_slice_size(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cmin_slice_size;
+}
+
+guint32
+gst_rtsp_wfd_client_get_video_slice_enc_params(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cslice_enc_params;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_framerate_control(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->cframe_rate_control;
+}
+
+guint32
+gst_rtsp_wfd_client_get_rtp_port0(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->crtp_port0;
+}
+
+guint32
+gst_rtsp_wfd_client_get_rtp_port1(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->crtp_port1;
+}
+
+gboolean
+gst_rtsp_wfd_client_get_edid_supported(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->edid_supported;
+}
+
+guint32
+gst_rtsp_wfd_client_get_edid_hresolution(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->edid_hres;
+}
+
+guint32
+gst_rtsp_wfd_client_get_edid_vresolution(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->edid_vres;
+}
+
+gboolean
+gst_rtsp_wfd_client_get_protection_enabled(GstRTSPWFDClient *client)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ return priv->protection_enabled;
+}
+
+void
+gst_rtsp_wfd_client_set_audio_freq(GstRTSPWFDClient *client, guint freq)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ priv->cFreq = freq;
+}
+
+void
+gst_rtsp_wfd_client_set_edid_supported(GstRTSPWFDClient *client, gboolean supported)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ priv->edid_supported = supported;
+}
+
+void
+gst_rtsp_wfd_client_set_edid_hresolution(GstRTSPWFDClient *client, guint32 reso)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ priv->edid_hres = reso;
+}
+
+void
+gst_rtsp_wfd_client_set_edid_vresolution(GstRTSPWFDClient *client, guint32 reso)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ priv->edid_vres = reso;
+}
+
+void
+gst_rtsp_wfd_client_set_protection_enabled(GstRTSPWFDClient *client, gboolean enable)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ priv->protection_enabled = enable;
+}
+
+void gst_rtsp_wfd_client_set_keep_alive_flag(GstRTSPWFDClient *client, gboolean flag)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_val_if_fail (priv != NULL, 0);
+
+ g_mutex_lock(&priv->keep_alive_lock);
+ if (priv->keep_alive_flag == !(flag))
+ priv->keep_alive_flag = flag;
+ g_mutex_unlock(&priv->keep_alive_lock);
+}
+
+void
+gst_rtsp_wfd_client_set_aud_codec (GstRTSPWFDClient *client, guint acodec)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->caCodec = acodec;
+}
+
+void
+gst_rtsp_wfd_client_set_audio_channels(GstRTSPWFDClient *client, guint channels)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cChanels = channels;
+}
+
+void
+gst_rtsp_wfd_client_set_audio_bit_width(GstRTSPWFDClient *client, guint bwidth)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cBitwidth = bwidth;
+}
+
+void
+gst_rtsp_wfd_client_set_audio_latency(GstRTSPWFDClient *client, guint latency)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->caLatency = latency;
+}
+
+void
+gst_rtsp_wfd_client_set_video_codec(GstRTSPWFDClient *client, guint vcodec)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cvCodec = vcodec;
+}
+
+void
+gst_rtsp_wfd_client_set_video_native(GstRTSPWFDClient *client, guint native)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cNative = native;
+}
+
+void
+gst_rtsp_wfd_client_set_vid_native_resolution(GstRTSPWFDClient *client, guint64 res)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cNativeResolution = res;
+}
+
+void
+gst_rtsp_wfd_client_set_video_cea_resolution(GstRTSPWFDClient *client, guint res)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cCEAResolution = res;
+}
+
+void
+gst_rtsp_wfd_client_set_video_vesa_resolution(GstRTSPWFDClient *client, guint res)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cVESAResolution = res;
+}
+
+void
+gst_rtsp_wfd_client_set_video_hh_resolution(GstRTSPWFDClient *client, guint res)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cHHResolution = res;
+}
+
+void
+gst_rtsp_wfd_client_set_video_profile(GstRTSPWFDClient *client, guint profile)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cProfile = profile;
+}
+
+void
+gst_rtsp_wfd_client_set_video_level(GstRTSPWFDClient *client, guint level)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cLevel = level;
+}
+
+void
+gst_rtsp_wfd_client_set_video_latency(GstRTSPWFDClient *client, guint latency)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cvLatency = latency;
+}
+
+void
+gst_rtsp_wfd_client_set_video_max_height(GstRTSPWFDClient *client, guint32 height)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cMaxHeight = height;
+}
+
+void
+gst_rtsp_wfd_client_set_video_max_width(GstRTSPWFDClient *client, guint32 width)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cMaxWidth = width;
+}
+
+void
+gst_rtsp_wfd_client_set_video_framerate(GstRTSPWFDClient *client, guint32 framerate)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cFramerate = framerate;
+}
+
+void
+gst_rtsp_wfd_client_set_video_min_slice_size(GstRTSPWFDClient *client, guint32 slice_size)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cmin_slice_size = slice_size;
+}
+
+void
+gst_rtsp_wfd_client_set_video_slice_enc_params(GstRTSPWFDClient *client, guint32 enc_params)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cslice_enc_params = enc_params;
+}
+
+void
+gst_rtsp_wfd_client_set_video_framerate_control(GstRTSPWFDClient *client, guint framerate)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->cframe_rate_control = framerate;
+}
+
+void
+gst_rtsp_wfd_client_set_rtp_port0(GstRTSPWFDClient *client, guint32 port)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->crtp_port0 = port;
+}
+
+void
+gst_rtsp_wfd_client_set_rtp_port1(GstRTSPWFDClient *client, guint32 port)
+{
+ GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+ g_return_if_fail (priv != NULL);
+
+ priv->crtp_port1 = port;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspconnection.h>
+
+#ifndef __GST_RTSP_WFD_CLIENT_H__
+#define __GST_RTSP_WFD_CLIENT_H__
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPWFDClient GstRTSPWFDClient;
+typedef struct _GstRTSPWFDClientClass GstRTSPWFDClientClass;
+typedef struct _GstRTSPWFDClientPrivate GstRTSPWFDClientPrivate;
+
+#include "rtsp-context.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-sdp.h"
+#include "rtsp-auth.h"
+#include "rtsp-client.h"
+#include "gstwfdmessage.h"
+
+#define GST_TYPE_RTSP_WFD_CLIENT (gst_rtsp_wfd_client_get_type ())
+#define GST_IS_RTSP_WFD_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_WFD_CLIENT))
+#define GST_IS_RTSP_WFD_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_WFD_CLIENT))
+#define GST_RTSP_WFD_CLIENT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_WFD_CLIENT, GstRTSPWFDClientClass))
+#define GST_RTSP_WFD_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_WFD_CLIENT, GstRTSPWFDClient))
+#define GST_RTSP_WFD_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_WFD_CLIENT, GstRTSPWFDClientClass))
+#define GST_RTSP_WFD_CLIENT_CAST(obj) ((GstRTSPWFDClient*)(obj))
+#define GST_RTSP_WFD_CLIENT_CLASS_CAST(klass) ((GstRTSPWFDClientClass*)(klass))
+
+
+/**
+ *
+ */
+typedef enum {
+ WFD_TRIGGER_SETUP,
+ WFD_TRIGGER_PAUSE,
+ WFD_TRIGGER_TEARDOWN,
+ WFD_TRIGGER_PLAY
+} GstWFDTriggerType;
+
+/**
+ * GstRTSPWFDClientSendFunc:
+ * @client: a #GstRTSPWFDClient
+ * @message: a #GstRTSPMessage
+ * @close: close the connection
+ * @user_data: user data when registering the callback
+ *
+ * This callback is called when @client wants to send @message. When @close is
+ * %TRUE, the connection should be closed when the message has been sent.
+ *
+ * Returns: %TRUE on success.
+ */
+typedef gboolean (*GstRTSPWFDClientSendFunc) (GstRTSPWFDClient *client,
+ GstRTSPMessage *message,
+ gboolean close,
+ gpointer user_data);
+
+/**
+ * GstRTSPWFDClient:
+ *
+ * The client object represents the connection and its state with a client.
+ */
+struct _GstRTSPWFDClient {
+ GstRTSPClient parent;
+
+ gint supported_methods;
+ /*< private >*/
+ GstRTSPWFDClientPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPWFDClientClass:
+ * @create_sdp: called when the SDP needs to be created for media.
+ * @configure_client_media: called when the stream in media needs to be configured.
+ * The default implementation will configure the blocksize on the payloader when
+ * spcified in the request headers.
+ * @configure_client_transport: called when the client transport needs to be
+ * configured.
+ * @params_set: set parameters. This function should also initialize the
+ * RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
+ * @params_get: get parameters. This function should also initialize the
+ * RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
+ *
+ * The client class structure.
+ */
+struct _GstRTSPWFDClientClass {
+ GstRTSPClientClass parent_class;
+
+ GstRTSPResult (*prepare_resource) (GstRTSPWFDClient *client, GstRTSPContext *ctx);
+ GstRTSPResult (*confirm_resource) (GstRTSPWFDClient *client, GstRTSPContext *ctx);
+
+ /* signals */
+ void (*wfd_options_request) (GstRTSPWFDClient *client, GstRTSPContext *ctx);
+ void (*wfd_get_param_request) (GstRTSPWFDClient *client, GstRTSPContext *ctx);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType gst_rtsp_wfd_client_get_type (void);
+
+GstRTSPWFDClient * gst_rtsp_wfd_client_new (void);
+
+void gst_rtsp_wfd_client_set_host_address (
+ GstRTSPWFDClient *client, const gchar * address);
+
+void gst_rtsp_wfd_client_start_wfd(GstRTSPWFDClient *client);
+GstRTSPResult gst_rtsp_wfd_client_trigger_request (
+ GstRTSPWFDClient * client, GstWFDTriggerType type);
+
+GstRTSPResult gst_rtsp_wfd_client_set_video_supported_resolution (
+ GstRTSPWFDClient * client, guint64 supported_reso);
+GstRTSPResult gst_rtsp_wfd_client_set_video_native_resolution (
+ GstRTSPWFDClient * client, guint64 native_reso);
+GstRTSPResult gst_rtsp_wfd_client_set_audio_codec (
+ GstRTSPWFDClient * client, guint8 audio_codec);
+
+guint gst_rtsp_wfd_client_get_audio_codec(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_audio_freq(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_audio_channels(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_audio_bit_width(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_audio_latency(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_video_codec(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_video_native(GstRTSPWFDClient *client);
+guint64 gst_rtsp_wfd_client_get_video_native_resolution(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_video_cea_resolution(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_video_vesa_resolution(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_video_hh_resolution(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_video_profile(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_video_level(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_video_latency(GstRTSPWFDClient *client);
+guint32 gst_rtsp_wfd_client_get_video_max_height(GstRTSPWFDClient *client);
+guint32 gst_rtsp_wfd_client_get_video_max_width(GstRTSPWFDClient *client);
+guint32 gst_rtsp_wfd_client_get_video_framerate(GstRTSPWFDClient *client);
+guint32 gst_rtsp_wfd_client_get_video_min_slice_size(GstRTSPWFDClient *client);
+guint32 gst_rtsp_wfd_client_get_video_slice_enc_params(GstRTSPWFDClient *client);
+guint gst_rtsp_wfd_client_get_video_framerate_control(GstRTSPWFDClient *client);
+guint32 gst_rtsp_wfd_client_get_rtp_port0(GstRTSPWFDClient *client);
+guint32 gst_rtsp_wfd_client_get_rtp_port1(GstRTSPWFDClient *client);
+gboolean gst_rtsp_wfd_client_get_edid_supported(GstRTSPWFDClient *client);
+guint32 gst_rtsp_wfd_client_get_edid_hresolution(GstRTSPWFDClient *client);
+guint32 gst_rtsp_wfd_client_get_edid_vresolution(GstRTSPWFDClient *client);
+gboolean gst_rtsp_wfd_client_get_protection_enabled(GstRTSPWFDClient *client);
+
+void gst_rtsp_wfd_client_set_audio_freq(GstRTSPWFDClient *client, guint freq);
+void gst_rtsp_wfd_client_set_edid_supported(GstRTSPWFDClient *client, gboolean supported);
+void gst_rtsp_wfd_client_set_edid_hresolution(GstRTSPWFDClient *client, guint32 reso);
+void gst_rtsp_wfd_client_set_edid_vresolution(GstRTSPWFDClient *client, guint32 reso);
+void gst_rtsp_wfd_client_set_protection_enabled(GstRTSPWFDClient *client, gboolean enable);
+void gst_rtsp_wfd_client_set_keep_alive_flag(GstRTSPWFDClient *client, gboolean flag);
+void gst_rtsp_wfd_client_set_aud_codec(GstRTSPWFDClient *client, guint acodec);
+void gst_rtsp_wfd_client_set_audio_channels(GstRTSPWFDClient *client, guint channels);
+void gst_rtsp_wfd_client_set_audio_bit_width(GstRTSPWFDClient *client, guint bwidth);
+void gst_rtsp_wfd_client_set_audio_latency(GstRTSPWFDClient *client, guint latency);
+void gst_rtsp_wfd_client_set_video_codec(GstRTSPWFDClient *client, guint vcodec);
+void gst_rtsp_wfd_client_set_video_native(GstRTSPWFDClient *client, guint native);
+void gst_rtsp_wfd_client_set_vid_native_resolution(GstRTSPWFDClient *client, guint64 res);
+void gst_rtsp_wfd_client_set_video_cea_resolution(GstRTSPWFDClient *client, guint res);
+void gst_rtsp_wfd_client_set_video_vesa_resolution(GstRTSPWFDClient *client, guint res);
+void gst_rtsp_wfd_client_set_video_hh_resolution(GstRTSPWFDClient *client, guint res);
+void gst_rtsp_wfd_client_set_video_profile(GstRTSPWFDClient *client, guint profile);
+void gst_rtsp_wfd_client_set_video_level(GstRTSPWFDClient *client, guint level);
+void gst_rtsp_wfd_client_set_video_latency(GstRTSPWFDClient *client, guint latency);
+void gst_rtsp_wfd_client_set_video_max_height(GstRTSPWFDClient *client, guint32 height);
+void gst_rtsp_wfd_client_set_video_max_width(GstRTSPWFDClient *client, guint32 width);
+void gst_rtsp_wfd_client_set_video_framerate(GstRTSPWFDClient *client, guint32 framerate);
+void gst_rtsp_wfd_client_set_video_min_slice_size(GstRTSPWFDClient *client, guint32 slice_size);
+void gst_rtsp_wfd_client_set_video_slice_enc_params(GstRTSPWFDClient *client, guint32 enc_params);
+void gst_rtsp_wfd_client_set_video_framerate_control(GstRTSPWFDClient *client, guint framerate);
+void gst_rtsp_wfd_client_set_rtp_port0(GstRTSPWFDClient *client, guint32 port);
+void gst_rtsp_wfd_client_set_rtp_port1(GstRTSPWFDClient *client, guint32 port);
+
+/**
+ * GstRTSPWFDClientSessionFilterFunc:
+ * @client: a #GstRTSPWFDClient object
+ * @sess: a #GstRTSPSession in @client
+ * @user_data: user data that has been given to gst_rtsp_wfd_client_session_filter()
+ *
+ * This function will be called by the gst_rtsp_wfd_client_session_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed
+ * from @client.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in
+ * @client.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of
+ * gst_rtsp_wfd_client_session_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPWFDClientSessionFilterFunc) (GstRTSPWFDClient *client,
+ GstRTSPSession *sess,
+ gpointer user_data);
+
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_WFD_CLIENT_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-client
+ * @short_description: A client connection state
+ * @see_also: #GstRTSPServer, #GstRTSPThreadPool
+ *
+ * The client object handles the connection with a client for as long as a TCP
+ * connection is open.
+ *
+ * A #GstRTSPClient is created by #GstRTSPServer when a new connection is
+ * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
+ * #GstRTSPAuth and #GstRTSPThreadPool from the server.
+ *
+ * The client connection should be configured with the #GstRTSPConnection using
+ * gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
+ * using gst_rtsp_client_attach(). From then on the client will handle requests
+ * on the connection.
+ *
+ * Use gst_rtsp_client_session_filter() to iterate or modify all the
+ * #GstRTSPSession objects managed by the client object.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include <stdio.h>
+#include <string.h>
+
+#include <gst/sdp/gstmikey.h>
+
+#include "rtsp-client.h"
+#include "rtsp-sdp.h"
+#include "rtsp-params.h"
+
+#define GST_RTSP_CLIENT_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
+
+/* locking order:
+ * send_lock, lock, tunnels_lock
+ */
+
+struct _GstRTSPClientPrivate
+{
+ GMutex lock; /* protects everything else */
+ GMutex send_lock;
+ GMutex watch_lock;
+ GstRTSPConnection *connection;
+ GstRTSPWatch *watch;
+ GMainContext *watch_context;
+ guint close_seq;
+ gchar *server_ip;
+ gboolean is_ipv6;
+
+ GstRTSPClientSendFunc send_func; /* protected by send_lock */
+ gpointer send_data; /* protected by send_lock */
+ GDestroyNotify send_notify; /* protected by send_lock */
+
+ GstRTSPSessionPool *session_pool;
+ gulong session_removed_id;
+ GstRTSPMountPoints *mount_points;
+ GstRTSPAuth *auth;
+ GstRTSPThreadPool *thread_pool;
+
+ /* used to cache the media in the last requested DESCRIBE so that
+ * we can pick it up in the next SETUP immediately */
+ gchar *path;
+ GstRTSPMedia *media;
+
+ GList *transports;
+ GList *sessions;
+ guint sessions_cookie;
+
+ gboolean drop_backlog;
+};
+
+static GMutex tunnels_lock;
+static GHashTable *tunnels; /* protected by tunnels_lock */
+
+#define DEFAULT_SESSION_POOL NULL
+#define DEFAULT_MOUNT_POINTS NULL
+#define DEFAULT_DROP_BACKLOG TRUE
+
+enum
+{
+ PROP_0,
+ PROP_SESSION_POOL,
+ PROP_MOUNT_POINTS,
+ PROP_DROP_BACKLOG,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_CLOSED,
+ SIGNAL_NEW_SESSION,
+ SIGNAL_OPTIONS_REQUEST,
+ SIGNAL_DESCRIBE_REQUEST,
+ SIGNAL_SETUP_REQUEST,
+ SIGNAL_PLAY_REQUEST,
+ SIGNAL_PAUSE_REQUEST,
+ SIGNAL_TEARDOWN_REQUEST,
+ SIGNAL_SET_PARAMETER_REQUEST,
+ SIGNAL_GET_PARAMETER_REQUEST,
+ SIGNAL_HANDLE_RESPONSE,
+ SIGNAL_SEND_MESSAGE,
+ SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
+#define GST_CAT_DEFAULT rtsp_client_debug
+
+static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
+
+static void gst_rtsp_client_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_client_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_client_finalize (GObject * obj);
+
+static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
+static void unlink_session_transports (GstRTSPClient * client,
+ GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
+static gboolean default_configure_client_media (GstRTSPClient * client,
+ GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
+static gboolean default_configure_client_transport (GstRTSPClient * client,
+ GstRTSPContext * ctx, GstRTSPTransport * ct);
+static GstRTSPResult default_params_set (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static GstRTSPResult default_params_get (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static gchar *default_make_path_from_uri (GstRTSPClient * client,
+ const GstRTSPUrl * uri);
+static gboolean default_handle_options_request (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static gboolean default_handle_set_param_request (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static gboolean default_handle_get_param_request (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static void client_session_removed (GstRTSPSessionPool * pool,
+ GstRTSPSession * session, GstRTSPClient * client);
+
+G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_client_class_init (GstRTSPClientClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_client_get_property;
+ gobject_class->set_property = gst_rtsp_client_set_property;
+ gobject_class->finalize = gst_rtsp_client_finalize;
+
+ klass->create_sdp = create_sdp;
+ klass->configure_client_media = default_configure_client_media;
+ klass->configure_client_transport = default_configure_client_transport;
+ klass->params_set = default_params_set;
+ klass->params_get = default_params_get;
+ klass->make_path_from_uri = default_make_path_from_uri;
+ klass->handle_options_request = default_handle_options_request;
+ klass->handle_set_param_request = default_handle_set_param_request;
+ klass->handle_get_param_request = default_handle_get_param_request;
+
+ g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
+ g_param_spec_object ("session-pool", "Session Pool",
+ "The session pool to use for client session",
+ GST_TYPE_RTSP_SESSION_POOL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
+ g_param_spec_object ("mount-points", "Mount Points",
+ "The mount points to use for client session",
+ GST_TYPE_RTSP_MOUNT_POINTS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
+ g_param_spec_boolean ("drop-backlog", "Drop Backlog",
+ "Drop data when the backlog queue is full",
+ DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_client_signals[SIGNAL_CLOSED] =
+ g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
+ g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+ gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
+ g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
+ g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
+
+ gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
+ g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
+
+ gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
+ g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
+
+ gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
+ g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
+
+ gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
+ g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
+
+ gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
+ g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
+
+ gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
+ g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
+
+ gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
+ g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ set_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
+ g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ get_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
+ g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ handle_response), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::send-message:
+ * @client: The RTSP client
+ * @session: (type GstRtspServer.RTSPSession): The session
+ * @message: (type GstRtsp.RTSPMessage): The message
+ */
+ gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
+ g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_POINTER);
+
+ tunnels =
+ g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
+ g_mutex_init (&tunnels_lock);
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
+}
+
+static void
+gst_rtsp_client_init (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
+
+ client->priv = priv;
+
+ g_mutex_init (&priv->lock);
+ g_mutex_init (&priv->send_lock);
+ g_mutex_init (&priv->watch_lock);
+ priv->close_seq = 0;
+ priv->drop_backlog = DEFAULT_DROP_BACKLOG;
+}
+
+static GstRTSPFilterResult
+filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
+ gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
+ unlink_session_transports (client, sess, sessmedia);
+
+ /* unmanage the media in the session */
+ return GST_RTSP_FILTER_REMOVE;
+}
+
+static void
+client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* check if we already know about this session */
+ if (g_list_find (priv->sessions, session) == NULL) {
+ GST_INFO ("watching session %p", session);
+
+ priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
+ priv->sessions_cookie++;
+
+ /* connect removed session handler, it will be disconnected when the last
+ * session gets removed */
+ if (priv->session_removed_id == 0)
+ priv->session_removed_id = g_signal_connect_data (priv->session_pool,
+ "session-removed", G_CALLBACK (client_session_removed),
+ g_object_ref (client), (GClosureNotify) g_object_unref, 0);
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return;
+}
+
+/* should be called with lock */
+static void
+client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
+ GList * link)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_INFO ("client %p: unwatch session %p", client, session);
+
+ if (link == NULL) {
+ link = g_list_find (priv->sessions, session);
+ if (link == NULL)
+ return;
+ }
+
+ priv->sessions = g_list_delete_link (priv->sessions, link);
+ priv->sessions_cookie++;
+
+ /* if this was the last session, disconnect the handler.
+ * This will also drop the extra client ref */
+ if (!priv->sessions) {
+ g_signal_handler_disconnect (priv->session_pool, priv->session_removed_id);
+ priv->session_removed_id = 0;
+ }
+
+ /* unlink all media managed in this session */
+ gst_rtsp_session_filter (session, filter_session_media, client);
+
+ /* remove the session */
+ g_object_unref (session);
+}
+
+static GstRTSPFilterResult
+cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
+ gpointer user_data)
+{
+ return GST_RTSP_FILTER_REMOVE;
+}
+
+/* A client is finalized when the connection is broken */
+static void
+gst_rtsp_client_finalize (GObject * obj)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (obj);
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_INFO ("finalize client %p", client);
+
+ if (priv->watch)
+ gst_rtsp_watch_set_flushing (priv->watch, TRUE);
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+
+ if (priv->watch)
+ g_source_destroy ((GSource *) priv->watch);
+
+ if (priv->watch_context)
+ g_main_context_unref (priv->watch_context);
+
+ /* all sessions should have been removed by now. We keep a ref to
+ * the client object for the session removed handler. The ref is
+ * dropped when the last session is removed from the list. */
+ g_assert (priv->sessions == NULL);
+ g_assert (priv->session_removed_id == 0);
+
+ if (priv->connection)
+ gst_rtsp_connection_free (priv->connection);
+ if (priv->session_pool) {
+ g_object_unref (priv->session_pool);
+ }
+ if (priv->mount_points)
+ g_object_unref (priv->mount_points);
+ if (priv->auth)
+ g_object_unref (priv->auth);
+ if (priv->thread_pool)
+ g_object_unref (priv->thread_pool);
+
+ if (priv->path)
+ g_free (priv->path);
+ if (priv->media) {
+ gst_rtsp_media_unprepare (priv->media);
+ g_object_unref (priv->media);
+ }
+
+ g_free (priv->server_ip);
+ g_mutex_clear (&priv->lock);
+ g_mutex_clear (&priv->send_lock);
+ g_mutex_clear (&priv->watch_lock);
+
+ G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_client_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (object);
+ GstRTSPClientPrivate *priv = client->priv;
+
+ switch (propid) {
+ case PROP_SESSION_POOL:
+ g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
+ break;
+ case PROP_MOUNT_POINTS:
+ g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
+ break;
+ case PROP_DROP_BACKLOG:
+ g_value_set_boolean (value, priv->drop_backlog);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_client_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (object);
+ GstRTSPClientPrivate *priv = client->priv;
+
+ switch (propid) {
+ case PROP_SESSION_POOL:
+ gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
+ break;
+ case PROP_MOUNT_POINTS:
+ gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
+ break;
+ case PROP_DROP_BACKLOG:
+ g_mutex_lock (&priv->lock);
+ priv->drop_backlog = g_value_get_boolean (value);
+ g_mutex_unlock (&priv->lock);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_client_new:
+ *
+ * Create a new #GstRTSPClient instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPClient
+ */
+GstRTSPClient *
+gst_rtsp_client_new (void)
+{
+ GstRTSPClient *result;
+
+ result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
+
+ return result;
+}
+
+static void
+send_message (GstRTSPClient * client, GstRTSPContext * ctx,
+ GstRTSPMessage * message, gboolean close)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
+ "GStreamer RTSP server");
+
+ /* remove any previous header */
+ gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
+
+ /* add the new session header for new session ids */
+ if (ctx->session) {
+ gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
+ gst_rtsp_session_get_header (ctx->session));
+ }
+
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (message);
+ }
+
+ if (close)
+ gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
+ 0, ctx, message);
+
+ g_mutex_lock (&priv->send_lock);
+ if (priv->send_func)
+ priv->send_func (client, message, close, priv->send_data);
+ g_mutex_unlock (&priv->send_lock);
+
+ gst_rtsp_message_unset (message);
+}
+
+static void
+send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
+ GstRTSPContext * ctx)
+{
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ ctx->session = NULL;
+
+ send_message (client, ctx, ctx->response, FALSE);
+}
+
+static void
+send_option_not_supported_response (GstRTSPClient * client,
+ GstRTSPContext * ctx, const gchar * unsupported_options)
+{
+ GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
+
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ if (unsupported_options != NULL) {
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
+ unsupported_options);
+ }
+
+ ctx->session = NULL;
+
+ send_message (client, ctx, ctx->response, FALSE);
+}
+
+static gboolean
+paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
+{
+ if (path1 == NULL || path2 == NULL)
+ return FALSE;
+
+ if (strlen (path1) != len2)
+ return FALSE;
+
+ if (strncmp (path1, path2, len2))
+ return FALSE;
+
+ return TRUE;
+}
+
+/* this function is called to initially find the media for the DESCRIBE request
+ * but is cached for when the same client (without breaking the connection) is
+ * doing a setup for the exact same url. */
+static GstRTSPMedia *
+find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
+ gint * matched)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ gint path_len;
+
+ /* find the longest matching factory for the uri first */
+ if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
+ path, matched)))
+ goto no_factory;
+
+ ctx->factory = factory;
+
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
+ goto no_factory_access;
+
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
+ goto not_authorized;
+
+ if (matched)
+ path_len = *matched;
+ else
+ path_len = strlen (path);
+
+ if (!paths_are_equal (priv->path, path, path_len)) {
+ GstRTSPThread *thread;
+
+ /* remove any previously cached values before we try to construct a new
+ * media for uri */
+ if (priv->path)
+ g_free (priv->path);
+ priv->path = NULL;
+ if (priv->media) {
+ gst_rtsp_media_unprepare (priv->media);
+ g_object_unref (priv->media);
+ }
+ priv->media = NULL;
+
+ /* prepare the media and add it to the pipeline */
+ if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
+ goto no_media;
+
+ ctx->media = media;
+
+ thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, ctx);
+ if (thread == NULL)
+ goto no_thread;
+
+ /* prepare the media */
+ if (!(gst_rtsp_media_prepare (media, thread)))
+ goto no_prepare;
+
+ /* now keep track of the uri and the media */
+ priv->path = g_strndup (path, path_len);
+ priv->media = media;
+ } else {
+ /* we have seen this path before, used cached media */
+ media = priv->media;
+ ctx->media = media;
+ GST_INFO ("reusing cached media %p for path %s", media, priv->path);
+ }
+
+ g_object_unref (factory);
+ ctx->factory = NULL;
+
+ if (media)
+ g_object_ref (media);
+
+ return media;
+
+ /* ERRORS */
+no_factory:
+ {
+ GST_ERROR ("client %p: no factory for path %s", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return NULL;
+ }
+no_factory_access:
+ {
+ GST_ERROR ("client %p: not authorized to see factory path %s", client,
+ path);
+ /* error reply is already sent */
+ return NULL;
+ }
+not_authorized:
+ {
+ GST_ERROR ("client %p: not authorized for factory path %s", client, path);
+ /* error reply is already sent */
+ return NULL;
+ }
+no_media:
+ {
+ GST_ERROR ("client %p: can't create media", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ g_object_unref (factory);
+ ctx->factory = NULL;
+ return NULL;
+ }
+no_thread:
+ {
+ GST_ERROR ("client %p: can't create thread", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_object_unref (media);
+ ctx->media = NULL;
+ g_object_unref (factory);
+ ctx->factory = NULL;
+ return NULL;
+ }
+no_prepare:
+ {
+ GST_ERROR ("client %p: can't prepare media", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_object_unref (media);
+ ctx->media = NULL;
+ g_object_unref (factory);
+ ctx->factory = NULL;
+ return NULL;
+ }
+}
+
+static gboolean
+do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPMessage message = { 0 };
+ GstMapInfo map_info;
+ guint8 *data;
+ guint usize;
+
+ gst_rtsp_message_init_data (&message, channel);
+
+ /* FIXME, need some sort of iovec RTSPMessage here */
+ if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
+ return FALSE;
+
+ gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
+
+ g_mutex_lock (&priv->send_lock);
+ if (priv->send_func)
+ priv->send_func (client, &message, FALSE, priv->send_data);
+ g_mutex_unlock (&priv->send_lock);
+
+ gst_rtsp_message_steal_body (&message, &data, &usize);
+ gst_buffer_unmap (buffer, &map_info);
+
+ gst_rtsp_message_unset (&message);
+
+ return TRUE;
+}
+
+static void
+link_transport (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPStreamTransport * trans)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_DEBUG ("client %p: linking transport %p", client, trans);
+
+ gst_rtsp_stream_transport_set_callbacks (trans,
+ (GstRTSPSendFunc) do_send_data,
+ (GstRTSPSendFunc) do_send_data, client, NULL);
+
+ priv->transports = g_list_prepend (priv->transports, trans);
+
+ /* make sure our session can't expire */
+ gst_rtsp_session_prevent_expire (session);
+}
+
+static void
+link_session_transports (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPSessionMedia * sessmedia)
+{
+ guint n_streams, i;
+
+ n_streams =
+ gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
+ for (i = 0; i < n_streams; i++) {
+ GstRTSPStreamTransport *trans;
+ const GstRTSPTransport *tr;
+
+ /* get the transport, if there is no transport configured, skip this stream */
+ trans = gst_rtsp_session_media_get_transport (sessmedia, i);
+ if (trans == NULL)
+ continue;
+
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+
+ if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+ /* for TCP, link the stream to the TCP connection of the client */
+ link_transport (client, session, trans);
+ }
+ }
+}
+
+static void
+unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPStreamTransport * trans)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_DEBUG ("client %p: unlinking transport %p", client, trans);
+
+ gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
+
+ priv->transports = g_list_remove (priv->transports, trans);
+
+ /* our session can now expire */
+ gst_rtsp_session_allow_expire (session);
+}
+
+static void
+unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPSessionMedia * sessmedia)
+{
+ guint n_streams, i;
+
+ n_streams =
+ gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
+ for (i = 0; i < n_streams; i++) {
+ GstRTSPStreamTransport *trans;
+ const GstRTSPTransport *tr;
+
+ /* get the transport, if there is no transport configured, skip this stream */
+ trans = gst_rtsp_session_media_get_transport (sessmedia, i);
+ if (trans == NULL)
+ continue;
+
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+
+ if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+ /* for TCP, unlink the stream from the TCP connection of the client */
+ unlink_transport (client, session, trans);
+ }
+ }
+}
+
+/**
+ * gst_rtsp_client_close:
+ * @client: a #GstRTSPClient
+ *
+ * Close the connection of @client and remove all media it was managing.
+ *
+ * Since: 1.4
+ */
+void
+gst_rtsp_client_close (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ const gchar *tunnelid;
+
+ GST_DEBUG ("client %p: closing connection", client);
+
+ if (priv->connection) {
+ if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
+ g_mutex_lock (&tunnels_lock);
+ /* remove from tunnelids */
+ g_hash_table_remove (tunnels, tunnelid);
+ g_mutex_unlock (&tunnels_lock);
+ }
+ gst_rtsp_connection_close (priv->connection);
+ }
+
+ /* connection is now closed, destroy the watch which will also cause the
+ * closed signal to be emitted */
+ if (priv->watch) {
+ GST_DEBUG ("client %p: destroying watch", client);
+ g_source_destroy ((GSource *) priv->watch);
+ priv->watch = NULL;
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ }
+}
+
+static gchar *
+default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
+{
+ gchar *path;
+
+ if (uri->query)
+ path = g_strconcat (uri->abspath, "?", uri->query, NULL);
+ else
+ path = g_strdup (uri->abspath);
+
+ return path;
+}
+
+static gboolean
+handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPClientClass *klass;
+ GstRTSPSession *session;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPStatusCode code;
+ gchar *path;
+ gint matched;
+ gboolean keep_session;
+
+ if (!ctx->session)
+ goto no_session;
+
+ session = ctx->session;
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, ctx->uri);
+
+ /* get a handle to the configuration of the media in the session */
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
+ goto not_found;
+
+ /* only aggregate control for now.. */
+ if (path[matched] != '\0')
+ goto no_aggregate;
+
+ g_free (path);
+
+ ctx->sessmedia = sessmedia;
+
+ /* we emit the signal before closing the connection */
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
+ 0, ctx);
+
+ /* make sure we unblock the backlog and don't accept new messages
+ * on the watch */
+ if (priv->watch != NULL)
+ gst_rtsp_watch_set_flushing (priv->watch, TRUE);
+
+ /* unlink the all TCP callbacks */
+ unlink_session_transports (client, session, sessmedia);
+
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
+
+ /* allow messages again so that we can send the reply */
+ if (priv->watch != NULL)
+ gst_rtsp_watch_set_flushing (priv->watch, FALSE);
+
+ /* unmanage the media in the session, returns false if all media session
+ * are torn down. */
+ keep_session = gst_rtsp_session_release_media (session, sessmedia);
+
+ /* construct the response now */
+ code = GST_RTSP_STS_OK;
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ send_message (client, ctx, ctx->response, TRUE);
+
+ if (!keep_session) {
+ /* remove the session */
+ gst_rtsp_session_pool_remove (priv->session_pool, session);
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+no_session:
+ {
+ GST_ERROR ("client %p: no session", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri supplied", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+not_found:
+ {
+ GST_ERROR ("client %p: no media for uri", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
+ return FALSE;
+ }
+no_aggregate:
+ {
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
+ return FALSE;
+ }
+}
+
+static GstRTSPResult
+default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res;
+
+ res = gst_rtsp_params_set (client, ctx);
+
+ return res;
+}
+
+static GstRTSPResult
+default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res;
+
+ res = gst_rtsp_params_get (client, ctx);
+
+ return res;
+}
+
+static gboolean
+default_handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res;
+ guint8 *data;
+ guint size;
+
+ res = gst_rtsp_message_get_body (ctx->request, &data, &size);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ if (size == 0) {
+ /* no body, keep-alive request */
+ send_generic_response (client, GST_RTSP_STS_OK, ctx);
+ } else {
+ /* there is a body, handle the params */
+ res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ send_message (client, ctx, ctx->response, FALSE);
+ }
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
+ 0, ctx);
+
+ return TRUE;
+
+ /* ERRORS */
+bad_request:
+ {
+ GST_ERROR ("client %p: bad request", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+}
+
+static gboolean
+default_handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res;
+ guint8 *data;
+ guint size;
+
+ res = gst_rtsp_message_get_body (ctx->request, &data, &size);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ if (size == 0) {
+ /* no body, keep-alive request */
+ send_generic_response (client, GST_RTSP_STS_OK, ctx);
+ } else {
+ /* there is a body, handle the params */
+ res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ send_message (client, ctx, ctx->response, FALSE);
+ }
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
+ 0, ctx);
+
+ return TRUE;
+
+ /* ERRORS */
+bad_request:
+ {
+ GST_ERROR ("client %p: bad request", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPSession *session;
+ GstRTSPClientClass *klass;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPStatusCode code;
+ GstRTSPState rtspstate;
+ gchar *path;
+ gint matched;
+
+ if (!(session = ctx->session))
+ goto no_session;
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, ctx->uri);
+
+ /* get a handle to the configuration of the media in the session */
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
+ goto not_found;
+
+ if (path[matched] != '\0')
+ goto no_aggregate;
+
+ g_free (path);
+
+ ctx->sessmedia = sessmedia;
+
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ /* the session state must be playing or recording */
+ if (rtspstate != GST_RTSP_STATE_PLAYING &&
+ rtspstate != GST_RTSP_STATE_RECORDING)
+ goto invalid_state;
+
+ /* unlink the all TCP callbacks */
+ unlink_session_transports (client, session, sessmedia);
+
+ /* then pause sending */
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
+
+ /* construct the response now */
+ code = GST_RTSP_STS_OK;
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ /* the state is now READY */
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
+
+ return TRUE;
+
+ /* ERRORS */
+no_session:
+ {
+ GST_ERROR ("client %p: no seesion", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri supplied", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+not_found:
+ {
+ GST_ERROR ("client %p: no media for uri", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
+ return FALSE;
+ }
+no_aggregate:
+ {
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
+ return FALSE;
+ }
+invalid_state:
+ {
+ GST_ERROR ("client %p: not PLAYING or RECORDING", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ ctx);
+ return FALSE;
+ }
+}
+
+/* convert @url and @path to a URL used as a content base for the factory
+ * located at @path */
+static gchar *
+make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
+{
+ GstRTSPUrl tmp;
+ gchar *result;
+ const gchar *trail;
+
+ /* check for trailing '/' and append one */
+ trail = (path[strlen (path) - 1] != '/' ? "/" : "");
+
+ tmp = *url;
+ tmp.user = NULL;
+ tmp.passwd = NULL;
+ tmp.abspath = g_strdup_printf ("%s%s", path, trail);
+ tmp.query = NULL;
+ result = gst_rtsp_url_get_request_uri (&tmp);
+ g_free (tmp.abspath);
+
+ return result;
+}
+
+static gboolean
+handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPSession *session;
+ GstRTSPClientClass *klass;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
+ GstRTSPStatusCode code;
+ GstRTSPUrl *uri;
+ gchar *str;
+ GstRTSPTimeRange *range;
+ GstRTSPResult res;
+ GstRTSPState rtspstate;
+ GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
+ gchar *path, *rtpinfo;
+ gint matched;
+
+ if (!(session = ctx->session))
+ goto no_session;
+
+ if (!(uri = ctx->uri))
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
+
+ /* get a handle to the configuration of the media in the session */
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
+ goto not_found;
+
+ if (path[matched] != '\0')
+ goto no_aggregate;
+
+ g_free (path);
+
+ ctx->sessmedia = sessmedia;
+ ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
+
+ /* the session state must be playing or ready */
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
+ goto invalid_state;
+
+ /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
+ if (!gst_rtsp_media_unsuspend (media))
+ goto unsuspend_failed;
+
+ /* parse the range header if we have one */
+ res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
+ if (res == GST_RTSP_OK) {
+ if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
+ /* we have a range, seek to the position */
+ unit = range->unit;
+ gst_rtsp_media_seek (media, range);
+ gst_rtsp_range_free (range);
+ }
+ }
+
+ /* link the all TCP callbacks */
+ link_session_transports (client, session, sessmedia);
+
+ /* grab RTPInfo from the media now */
+ rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
+
+ /* construct the response now */
+ code = GST_RTSP_STS_OK;
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ /* add the RTP-Info header */
+ if (rtpinfo)
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
+ rtpinfo);
+
+ /* add the range */
+ str = gst_rtsp_media_get_range_string (media, TRUE, unit);
+ if (str)
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ /* start playing after sending the response */
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
+
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
+
+ return TRUE;
+
+ /* ERRORS */
+no_session:
+ {
+ GST_ERROR ("client %p: no session", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri supplied", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+not_found:
+ {
+ GST_ERROR ("client %p: media not found", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_aggregate:
+ {
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
+ return FALSE;
+ }
+invalid_state:
+ {
+ GST_ERROR ("client %p: not PLAYING or READY", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ ctx);
+ return FALSE;
+ }
+unsuspend_failed:
+ {
+ GST_ERROR ("client %p: unsuspend failed", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ return FALSE;
+ }
+}
+
+static void
+do_keepalive (GstRTSPSession * session)
+{
+ GST_INFO ("keep session %p alive", session);
+ gst_rtsp_session_touch (session);
+}
+
+/* parse @transport and return a valid transport in @tr. only transports
+ * supported by @stream are returned. Returns FALSE if no valid transport
+ * was found. */
+static gboolean
+parse_transport (const char *transport, GstRTSPStream * stream,
+ GstRTSPTransport * tr)
+{
+ gint i;
+ gboolean res;
+ gchar **transports;
+
+ res = FALSE;
+ gst_rtsp_transport_init (tr);
+
+ GST_DEBUG ("parsing transports %s", transport);
+
+ transports = g_strsplit (transport, ",", 0);
+
+ /* loop through the transports, try to parse */
+ for (i = 0; transports[i]; i++) {
+ res = gst_rtsp_transport_parse (transports[i], tr);
+ if (res != GST_RTSP_OK) {
+ /* no valid transport, search some more */
+ GST_WARNING ("could not parse transport %s", transports[i]);
+ goto next;
+ }
+
+ /* we have a transport, see if it's supported */
+ if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
+ GST_WARNING ("unsupported transport %s", transports[i]);
+ goto next;
+ }
+
+ /* we have a valid transport */
+ GST_INFO ("found valid transport %s", transports[i]);
+ res = TRUE;
+ break;
+
+ next:
+ gst_rtsp_transport_init (tr);
+ }
+ g_strfreev (transports);
+
+ return res;
+}
+
+static gboolean
+default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+ GstRTSPStream * stream, GstRTSPContext * ctx)
+{
+ GstRTSPMessage *request = ctx->request;
+ gchar *blocksize_str;
+
+ if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
+ &blocksize_str, 0) == GST_RTSP_OK) {
+ guint64 blocksize;
+ gchar *end;
+
+ blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
+ if (end == blocksize_str)
+ goto parse_failed;
+
+ /* we don't want to change the mtu when this media
+ * can be shared because it impacts other clients */
+ if (gst_rtsp_media_is_shared (media))
+ goto done;
+
+ if (blocksize > G_MAXUINT)
+ blocksize = G_MAXUINT;
+
+ gst_rtsp_stream_set_mtu (stream, blocksize);
+ }
+done:
+ return TRUE;
+
+ /* ERRORS */
+parse_failed:
+ {
+ GST_ERROR_OBJECT (client, "failed to parse blocksize");
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+}
+
+static gboolean
+default_configure_client_transport (GstRTSPClient * client,
+ GstRTSPContext * ctx, GstRTSPTransport * ct)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ /* we have a valid transport now, set the destination of the client. */
+ if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ gboolean use_client_settings;
+
+ use_client_settings =
+ gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
+
+ if (ct->destination && use_client_settings) {
+ GstRTSPAddress *addr;
+
+ addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
+ ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
+
+ if (addr == NULL)
+ goto no_address;
+
+ gst_rtsp_address_free (addr);
+ } else {
+ GstRTSPAddress *addr;
+ GSocketFamily family;
+
+ family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
+
+ addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
+ if (addr == NULL)
+ goto no_address;
+
+ g_free (ct->destination);
+ ct->destination = g_strdup (addr->address);
+ ct->port.min = addr->port;
+ ct->port.max = addr->port + addr->n_ports - 1;
+ ct->ttl = addr->ttl;
+
+ gst_rtsp_address_free (addr);
+ }
+ } else {
+ GstRTSPUrl *url;
+
+ url = gst_rtsp_connection_get_url (priv->connection);
+ g_free (ct->destination);
+ ct->destination = g_strdup (url->host);
+
+ if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
+ GSocket *sock;
+ GSocketAddress *addr;
+
+ sock = gst_rtsp_connection_get_read_socket (priv->connection);
+ if ((addr = g_socket_get_remote_address (sock, NULL))) {
+ /* our read port is the sender port of client */
+ ct->client_port.min =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ if ((addr = g_socket_get_local_address (sock, NULL))) {
+ ct->server_port.max =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ sock = gst_rtsp_connection_get_write_socket (priv->connection);
+ if ((addr = g_socket_get_remote_address (sock, NULL))) {
+ /* our write port is the receiver port of client */
+ ct->client_port.max =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ if ((addr = g_socket_get_local_address (sock, NULL))) {
+ ct->server_port.min =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ /* check if the client selected channels for TCP */
+ if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
+ gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
+ &ct->interleaved);
+ }
+ }
+ }
+ return TRUE;
+
+ /* ERRORS */
+no_address:
+ {
+ GST_ERROR_OBJECT (client, "failed to acquire address for stream");
+ return FALSE;
+ }
+}
+
+static GstRTSPTransport *
+make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
+ GstRTSPTransport * ct)
+{
+ GstRTSPTransport *st;
+ GInetAddress *addr;
+ GSocketFamily family;
+
+ /* prepare the server transport */
+ gst_rtsp_transport_new (&st);
+
+ st->trans = ct->trans;
+ st->profile = ct->profile;
+ st->lower_transport = ct->lower_transport;
+
+ addr = g_inet_address_new_from_string (ct->destination);
+
+ if (!addr) {
+ GST_ERROR ("failed to get inet addr from client destination");
+ family = G_SOCKET_FAMILY_IPV4;
+ } else {
+ family = g_inet_address_get_family (addr);
+ g_object_unref (addr);
+ addr = NULL;
+ }
+
+ switch (st->lower_transport) {
+ case GST_RTSP_LOWER_TRANS_UDP:
+ st->client_port = ct->client_port;
+ gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
+ break;
+ case GST_RTSP_LOWER_TRANS_UDP_MCAST:
+ st->port = ct->port;
+ st->destination = g_strdup (ct->destination);
+ st->ttl = ct->ttl;
+ break;
+ case GST_RTSP_LOWER_TRANS_TCP:
+ st->interleaved = ct->interleaved;
+ st->client_port = ct->client_port;
+ st->server_port = ct->server_port;
+ default:
+ break;
+ }
+
+ gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
+
+ return st;
+}
+
+#define AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
+static gboolean
+mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
+{
+ const gchar *srtp_cipher;
+ const gchar *srtp_auth;
+ const GstMIKEYPayload *sp;
+ guint i;
+
+ /* loop over Security policy until we find one containing policy */
+ for (i = 0;; i++) {
+ if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
+ break;
+
+ if (((GstMIKEYPayloadSP *) sp)->policy == policy)
+ break;
+ }
+
+ /* the default ciphers */
+ srtp_cipher = "aes-128-icm";
+ srtp_auth = "hmac-sha1-80";
+
+ /* now override the defaults with what is in the Security Policy */
+ if (sp != NULL) {
+ guint len;
+
+ /* collect all the params and go over them */
+ len = gst_mikey_payload_sp_get_n_params (sp);
+ for (i = 0; i < len; i++) {
+ const GstMIKEYPayloadSPParam *param =
+ gst_mikey_payload_sp_get_param (sp, i);
+
+ switch (param->type) {
+ case GST_MIKEY_SP_SRTP_ENC_ALG:
+ switch (param->val[0]) {
+ case 0:
+ srtp_cipher = "null";
+ break;
+ case 2:
+ case 1:
+ srtp_cipher = "aes-128-icm";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
+ switch (param->val[0]) {
+ case AES_128_KEY_LEN:
+ srtp_cipher = "aes-128-icm";
+ break;
+ case AES_256_KEY_LEN:
+ srtp_cipher = "aes-256-icm";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_AUTH_ALG:
+ switch (param->val[0]) {
+ case 0:
+ srtp_auth = "null";
+ break;
+ case 2:
+ case 1:
+ srtp_auth = "hmac-sha1-80";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
+ switch (param->val[0]) {
+ case HMAC_32_KEY_LEN:
+ srtp_auth = "hmac-sha1-32";
+ break;
+ case HMAC_80_KEY_LEN:
+ srtp_auth = "hmac-sha1-80";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_SRTP_ENC:
+ break;
+ case GST_MIKEY_SP_SRTP_SRTCP_ENC:
+ break;
+ default:
+ break;
+ }
+ }
+ }
+ /* now configure the SRTP parameters */
+ gst_caps_set_simple (caps,
+ "srtp-cipher", G_TYPE_STRING, srtp_cipher,
+ "srtp-auth", G_TYPE_STRING, srtp_auth,
+ "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
+ "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
+
+ return TRUE;
+}
+
+static gboolean
+handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
+ guint8 * data, gsize size)
+{
+ GstMIKEYMessage *msg;
+ guint i, n_cs;
+ GstCaps *caps = NULL;
+ GstMIKEYPayloadKEMAC *kemac;
+ const GstMIKEYPayloadKeyData *pkd;
+ GstBuffer *key;
+
+ /* the MIKEY message contains a CSB or crypto session bundle. It is a
+ * set of Crypto Sessions protected with the same master key.
+ * In the context of SRTP, an RTP and its RTCP stream is part of a
+ * crypto session */
+ if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
+ goto parse_failed;
+
+ /* we can only handle SRTP crypto sessions for now */
+ if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
+ goto invalid_map_type;
+
+ /* get the number of crypto sessions. This maps SSRC to its
+ * security parameters */
+ n_cs = gst_mikey_message_get_n_cs (msg);
+ if (n_cs == 0)
+ goto no_crypto_sessions;
+
+ /* we also need keys */
+ if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
+ (msg, GST_MIKEY_PT_KEMAC, 0)))
+ goto no_keys;
+
+ /* we don't support encrypted keys */
+ if (kemac->enc_alg != GST_MIKEY_ENC_NULL
+ || kemac->mac_alg != GST_MIKEY_MAC_NULL)
+ goto unsupported_encryption;
+
+ /* get Key data sub-payload */
+ pkd = (const GstMIKEYPayloadKeyData *)
+ gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
+
+ key =
+ gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
+ pkd->key_len);
+
+ /* go over all crypto sessions and create the security policy for each
+ * SSRC */
+ for (i = 0; i < n_cs; i++) {
+ const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
+
+ caps = gst_caps_new_simple ("application/x-srtp",
+ "ssrc", G_TYPE_UINT, map->ssrc,
+ "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
+ mikey_apply_policy (caps, msg, map->policy);
+
+ gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
+ gst_caps_unref (caps);
+ }
+ gst_mikey_message_unref (msg);
+ gst_buffer_unref (key);
+
+ return TRUE;
+
+ /* ERRORS */
+parse_failed:
+ {
+ GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
+ return FALSE;
+ }
+invalid_map_type:
+ {
+ GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
+ goto cleanup_message;
+ }
+no_crypto_sessions:
+ {
+ GST_DEBUG_OBJECT (client, "no crypto sessions");
+ goto cleanup_message;
+ }
+no_keys:
+ {
+ GST_DEBUG_OBJECT (client, "no keys found");
+ goto cleanup_message;
+ }
+unsupported_encryption:
+ {
+ GST_DEBUG_OBJECT (client, "unsupported key encryption");
+ goto cleanup_message;
+ }
+cleanup_message:
+ {
+ gst_mikey_message_unref (msg);
+ return FALSE;
+ }
+}
+
+#define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
+
+static void
+strip_chars (gchar * str)
+{
+ gchar *s;
+ gsize len;
+
+ len = strlen (str);
+ while (len--) {
+ if (!IS_STRIP_CHAR (str[len]))
+ break;
+ str[len] = '\0';
+ }
+ for (s = str; *s && IS_STRIP_CHAR (*s); s++);
+ memmove (str, s, len + 1);
+}
+
+/* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
+ * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
+ */
+static gboolean
+handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
+{
+ gchar **specs;
+ gint i, j;
+
+ specs = g_strsplit (keymgmt, ",", 0);
+ for (i = 0; specs[i]; i++) {
+ gchar **split;
+
+ split = g_strsplit (specs[i], ";", 0);
+ for (j = 0; split[j]; j++) {
+ g_strstrip (split[j]);
+ if (g_str_has_prefix (split[j], "prot=")) {
+ g_strstrip (split[j] + 5);
+ if (!g_str_equal (split[j] + 5, "mikey"))
+ break;
+ GST_DEBUG ("found mikey");
+ } else if (g_str_has_prefix (split[j], "uri=")) {
+ strip_chars (split[j] + 4);
+ GST_DEBUG ("found uri '%s'", split[j] + 4);
+ } else if (g_str_has_prefix (split[j], "data=")) {
+ guchar *data;
+ gsize size;
+ strip_chars (split[j] + 5);
+ GST_DEBUG ("found data '%s'", split[j] + 5);
+ data = g_base64_decode_inplace (split[j] + 5, &size);
+ handle_mikey_data (client, ctx, data, size);
+ }
+ }
+ g_strfreev (split);
+ }
+ g_strfreev (specs);
+ return TRUE;
+}
+
+static gboolean
+handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ GstRTSPUrl *uri;
+ gchar *transport, *keymgmt;
+ GstRTSPTransport *ct, *st;
+ GstRTSPStatusCode code;
+ GstRTSPSession *session;
+ GstRTSPStreamTransport *trans;
+ gchar *trans_str;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
+ GstRTSPStream *stream;
+ GstRTSPState rtspstate;
+ GstRTSPClientClass *klass;
+ gchar *path, *control;
+ gint matched;
+ gboolean new_session = FALSE;
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ uri = ctx->uri;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
+
+ /* parse the transport */
+ res =
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
+ &transport, 0);
+ if (res != GST_RTSP_OK)
+ goto no_transport;
+
+ /* we create the session after parsing stuff so that we don't make
+ * a session for malformed requests */
+ if (priv->session_pool == NULL)
+ goto no_pool;
+
+ session = ctx->session;
+
+ if (session) {
+ g_object_ref (session);
+ /* get a handle to the configuration of the media in the session, this can
+ * return NULL if this is a new url to manage in this session. */
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ } else {
+ /* we need a new media configuration in this session */
+ sessmedia = NULL;
+ }
+
+ /* we have no session media, find one and manage it */
+ if (sessmedia == NULL) {
+ /* get a handle to the configuration of the media in the session */
+ media = find_media (client, ctx, path, &matched);
+ } else {
+ if ((media = gst_rtsp_session_media_get_media (sessmedia)))
+ g_object_ref (media);
+ else
+ goto media_not_found;
+ }
+ /* no media, not found then */
+ if (media == NULL)
+ goto media_not_found_no_reply;
+
+ /* FIXME-WFD : wfd url problem */
+#if 0
+ if (path[matched] == '\0')
+ goto control_not_found;
+
+ /* path is what matched. */
+ path[matched] = '\0';
+ /* control is remainder */
+ control = &path[matched + 1];
+#else
+ control = g_strdup ("stream=0");
+#endif
+
+ /* find the stream now using the control part */
+ stream = gst_rtsp_media_find_stream (media, control);
+ if (stream == NULL)
+ goto stream_not_found;
+
+ /* now we have a uri identifying a valid media and stream */
+ ctx->stream = stream;
+ ctx->media = media;
+
+ if (session == NULL) {
+ /* create a session if this fails we probably reached our session limit or
+ * something. */
+ if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
+ goto service_unavailable;
+
+ /* make sure this client is closed when the session is closed */
+ client_watch_session (client, session);
+
+ new_session = TRUE;
+ /* signal new session */
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
+ session);
+
+ ctx->session = session;
+ }
+
+ if (!klass->configure_client_media (client, media, stream, ctx))
+ goto configure_media_failed_no_reply;
+
+ gst_rtsp_transport_new (&ct);
+
+ /* parse and find a usable supported transport */
+ if (!parse_transport (transport, stream, ct))
+ goto unsupported_transports;
+
+ /* update the client transport */
+ if (!klass->configure_client_transport (client, ctx, ct))
+ goto unsupported_client_transport;
+
+ /* parse the keymgmt */
+ if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
+ &keymgmt, 0) == GST_RTSP_OK) {
+ if (!handle_keymgmt (client, ctx, keymgmt))
+ goto keymgmt_error;
+ }
+
+ if (sessmedia == NULL) {
+ /* manage the media in our session now, if not done already */
+ sessmedia = gst_rtsp_session_manage_media (session, path, media);
+ /* if we stil have no media, error */
+ if (sessmedia == NULL)
+ goto sessmedia_unavailable;
+ } else {
+ g_object_unref (media);
+ }
+
+ ctx->sessmedia = sessmedia;
+
+ /* set in the session media transport */
+ trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
+
+ /* configure the url used to set this transport, this we will use when
+ * generating the response for the PLAY request */
+ gst_rtsp_stream_transport_set_url (trans, uri);
+
+ /* configure keepalive for this transport */
+ gst_rtsp_stream_transport_set_keepalive (trans,
+ (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
+
+ /* create and serialize the server transport */
+ st = make_server_transport (client, ctx, ct);
+ trans_str = gst_rtsp_transport_as_text (st);
+
+ /* FIXME-WFD : Temporarily force to set profile string */
+ trans_str = g_strjoinv ("RTP/AVP/UDP", g_strsplit (trans_str, "RTP/AVP", -1));
+
+ gst_rtsp_transport_free (st);
+
+ /* construct the response now */
+ code = GST_RTSP_STS_OK;
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
+ trans_str);
+ g_free (trans_str);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ /* update the state */
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ switch (rtspstate) {
+ case GST_RTSP_STATE_PLAYING:
+ case GST_RTSP_STATE_RECORDING:
+ case GST_RTSP_STATE_READY:
+ /* no state change */
+ break;
+ default:
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
+ break;
+ }
+ g_object_unref (session);
+ g_free (path);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
+
+ return TRUE;
+
+ /* ERRORS */
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_transport:
+ {
+ GST_ERROR ("client %p: no transport", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ goto cleanup_path;
+ }
+no_pool:
+ {
+ GST_ERROR ("client %p: no session pool configured", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ goto cleanup_path;
+ }
+media_not_found_no_reply:
+ {
+ GST_ERROR ("client %p: media '%s' not found", client, path);
+ /* error reply is already sent */
+ goto cleanup_path;
+ }
+media_not_found:
+ {
+ GST_ERROR ("client %p: media '%s' not found", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ goto cleanup_path;
+ }
+#if 0
+control_not_found:
+ {
+ GST_ERROR ("client %p: no control in path '%s'", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_object_unref (media);
+ goto cleanup_path;
+ }
+#endif
+stream_not_found:
+ {
+ GST_ERROR ("client %p: stream '%s' not found", client, control);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_object_unref (media);
+ goto cleanup_path;
+ }
+service_unavailable:
+ {
+ GST_ERROR ("client %p: can't create session", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_object_unref (media);
+ goto cleanup_path;
+ }
+sessmedia_unavailable:
+ {
+ GST_ERROR ("client %p: can't create session media", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_object_unref (media);
+ goto cleanup_session;
+ }
+configure_media_failed_no_reply:
+ {
+ GST_ERROR ("client %p: configure_media failed", client);
+ /* error reply is already sent */
+ goto cleanup_session;
+ }
+unsupported_transports:
+ {
+ GST_ERROR ("client %p: unsupported transports", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ goto cleanup_transport;
+ }
+unsupported_client_transport:
+ {
+ GST_ERROR ("client %p: unsupported client transport", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ goto cleanup_transport;
+ }
+keymgmt_error:
+ {
+ GST_ERROR ("client %p: keymgmt error", client);
+ send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
+ goto cleanup_transport;
+ }
+ {
+ cleanup_transport:
+ gst_rtsp_transport_free (ct);
+ cleanup_session:
+ if (new_session)
+ gst_rtsp_session_pool_remove (priv->session_pool, session);
+ g_object_unref (session);
+ cleanup_path:
+ g_free (path);
+ return FALSE;
+ }
+}
+
+static GstSDPMessage *
+create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstSDPMessage *sdp;
+ GstSDPInfo info;
+ const gchar *proto;
+
+ gst_sdp_message_new (&sdp);
+
+ /* some standard things first */
+ gst_sdp_message_set_version (sdp, "0");
+
+ if (priv->is_ipv6)
+ proto = "IP6";
+ else
+ proto = "IP4";
+
+ gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
+ priv->server_ip);
+
+ gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
+ gst_sdp_message_set_information (sdp, "rtsp-server");
+ gst_sdp_message_add_time (sdp, "0", "0", NULL);
+ gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
+ gst_sdp_message_add_attribute (sdp, "type", "broadcast");
+ gst_sdp_message_add_attribute (sdp, "control", "*");
+
+ info.is_ipv6 = priv->is_ipv6;
+ info.server_ip = priv->server_ip;
+
+ /* create an SDP for the media object */
+ if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
+ goto no_sdp;
+
+ return sdp;
+
+ /* ERRORS */
+no_sdp:
+ {
+ GST_ERROR ("client %p: could not create SDP", client);
+ gst_sdp_message_free (sdp);
+ return NULL;
+ }
+}
+
+/* for the describe we must generate an SDP */
+static gboolean
+handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ GstSDPMessage *sdp;
+ guint i;
+ gchar *path, *str;
+ GstRTSPMedia *media;
+ GstRTSPClientClass *klass;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ /* check what kind of format is accepted, we don't really do anything with it
+ * and always return SDP for now. */
+ for (i = 0;; i++) {
+ gchar *accept;
+
+ res =
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
+ &accept, i);
+ if (res == GST_RTSP_ENOTIMPL)
+ break;
+
+ if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
+ break;
+ }
+
+ if (!priv->mount_points)
+ goto no_mount_points;
+
+ if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
+ goto no_path;
+
+ /* find the media object for the uri */
+ if (!(media = find_media (client, ctx, path, NULL)))
+ goto no_media;
+
+ /* create an SDP for the media object on this client */
+ if (!(sdp = klass->create_sdp (client, media)))
+ goto no_sdp;
+
+ /* we suspend after the describe */
+ gst_rtsp_media_suspend (media);
+ g_object_unref (media);
+
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/sdp");
+
+ /* content base for some clients that might screw up creating the setup uri */
+ str = make_base_url (client, ctx->uri, path);
+ g_free (path);
+
+ GST_INFO ("adding content-base: %s", str);
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
+
+ /* add SDP to the response body */
+ str = gst_sdp_message_as_text (sdp);
+ gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
+ gst_sdp_message_free (sdp);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
+ 0, ctx);
+
+ return TRUE;
+
+ /* ERRORS */
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_mount_points:
+ {
+ GST_ERROR ("client %p: no mount points configured", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_path:
+ {
+ GST_ERROR ("client %p: can't find path for url", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_media:
+ {
+ GST_ERROR ("client %p: no media", client);
+ g_free (path);
+ /* error reply is already sent */
+ return FALSE;
+ }
+no_sdp:
+ {
+ GST_ERROR ("client %p: can't create SDP", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_free (path);
+ g_object_unref (media);
+ return FALSE;
+ }
+}
+
+static gboolean
+default_handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPMethod options;
+ gchar *str;
+
+ options = GST_RTSP_DESCRIBE |
+ GST_RTSP_OPTIONS |
+ GST_RTSP_PAUSE |
+ GST_RTSP_PLAY |
+ GST_RTSP_SETUP |
+ GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
+
+ str = gst_rtsp_options_as_text (options);
+
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
+ g_free (str);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
+ 0, ctx);
+
+ return TRUE;
+}
+
+/* remove duplicate and trailing '/' */
+static void
+sanitize_uri (GstRTSPUrl * uri)
+{
+ gint i, len;
+ gchar *s, *d;
+ gboolean have_slash, prev_slash;
+
+ s = d = uri->abspath;
+ len = strlen (uri->abspath);
+
+ prev_slash = FALSE;
+
+ for (i = 0; i < len; i++) {
+ have_slash = s[i] == '/';
+ *d = s[i];
+ if (!have_slash || !prev_slash)
+ d++;
+ prev_slash = have_slash;
+ }
+ len = d - uri->abspath;
+ /* don't remove the first slash if that's the only thing left */
+ if (len > 1 && *(d - 1) == '/')
+ d--;
+ *d = '\0';
+}
+
+/* is called when the session is removed from its session pool. */
+static void
+client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
+ GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_INFO ("client %p: session %p removed", client, session);
+
+ g_mutex_lock (&priv->lock);
+ client_unwatch_session (client, session, NULL);
+ g_mutex_unlock (&priv->lock);
+}
+
+/* Returns TRUE if there are no Require headers, otherwise returns FALSE
+ * and also returns a newly-allocated string of (comma-separated) unsupported
+ * options in the unsupported_reqs variable .
+ *
+ * There may be multiple Require headers, but we must send one single
+ * Unsupported header with all the unsupported options as response. If
+ * an incoming Require header contained a comma-separated list of options
+ * GstRtspConnection will already have split that list up into multiple
+ * headers.
+ *
+ * TODO: allow the application to decide what features are supported
+ */
+static gboolean
+check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
+{
+ GstRTSPResult res;
+ GPtrArray *arr = NULL;
+ gchar *reqs = NULL;
+ gint i;
+
+ i = 0;
+ do {
+ res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
+
+ if (res == GST_RTSP_ENOTIMPL)
+ break;
+
+ if (arr == NULL)
+ arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
+
+ g_ptr_array_add (arr, g_strdup (reqs));
+ }
+ while (TRUE);
+
+ /* if we don't have any Require headers at all, all is fine */
+ if (i == 1)
+ return TRUE;
+
+ /* otherwise we've now processed at all the Require headers */
+ g_ptr_array_add (arr, NULL);
+
+ /* for now we don't commit to supporting anything, so will just report
+ * all of the required options as unsupported */
+ *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
+
+ g_ptr_array_unref (arr);
+ return FALSE;
+}
+
+static void
+handle_request (GstRTSPClient * client, GstRTSPMessage * request)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPMethod method;
+ const gchar *uristr;
+ GstRTSPUrl *uri = NULL;
+ GstRTSPVersion version;
+ GstRTSPResult res;
+ GstRTSPSession *session = NULL;
+ GstRTSPContext sctx = { NULL }, *ctx;
+ GstRTSPMessage response = { 0 };
+ gchar *unsupported_reqs = NULL;
+ gchar *sessid;
+ GstRTSPClientClass *klass;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->request = request;
+ ctx->response = &response;
+
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (request);
+ }
+
+ gst_rtsp_message_parse_request (request, &method, &uristr, &version);
+
+ GST_INFO ("client %p: received a request %s %s %s", client,
+ gst_rtsp_method_as_text (method), uristr,
+ gst_rtsp_version_as_text (version));
+
+ /* we can only handle 1.0 requests */
+ if (version != GST_RTSP_VERSION_1_0)
+ goto not_supported;
+
+ ctx->method = method;
+
+ /* we always try to parse the url first */
+ if (strcmp (uristr, "*") == 0) {
+ /* special case where we have * as uri, keep uri = NULL */
+ } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
+ /* check if the uristr is an absolute path <=> scheme and host information
+ * is missing */
+ gchar *scheme;
+
+ scheme = g_uri_parse_scheme (uristr);
+ if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
+ gchar *absolute_uristr = NULL;
+
+ GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
+ if (priv->server_ip == NULL) {
+ GST_WARNING_OBJECT (client, "host information missing");
+ goto bad_request;
+ }
+
+ absolute_uristr =
+ g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
+
+ GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
+ if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
+ g_free (absolute_uristr);
+ goto bad_request;
+ }
+ g_free (absolute_uristr);
+ } else {
+ g_free (scheme);
+ goto bad_request;
+ }
+ }
+
+ /* get the session if there is any */
+ res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
+ if (res == GST_RTSP_OK) {
+ if (priv->session_pool == NULL)
+ goto no_pool;
+
+ /* we had a session in the request, find it again */
+ if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
+ goto session_not_found;
+
+ /* we add the session to the client list of watched sessions. When a session
+ * disappears because it times out, we will be notified. If all sessions are
+ * gone, we will close the connection */
+ client_watch_session (client, session);
+ }
+
+ /* sanitize the uri */
+ if (uri)
+ sanitize_uri (uri);
+ ctx->uri = uri;
+ ctx->session = session;
+
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
+ goto not_authorized;
+
+#if 0
+ /* FIXME-WFD : How does it handle this */
+ /* handle any 'Require' headers */
+ if (!check_request_requirements (ctx->request, &unsupported_reqs))
+ goto unsupported_requirement;
+#endif
+
+ /* now see what is asked and dispatch to a dedicated handler */
+ switch (method) {
+ case GST_RTSP_OPTIONS:
+ klass->handle_options_request (client, ctx);
+ break;
+ case GST_RTSP_DESCRIBE:
+ handle_describe_request (client, ctx);
+ break;
+ case GST_RTSP_SETUP:
+ handle_setup_request (client, ctx);
+ break;
+ case GST_RTSP_PLAY:
+ handle_play_request (client, ctx);
+ break;
+ case GST_RTSP_PAUSE:
+ handle_pause_request (client, ctx);
+ break;
+ case GST_RTSP_TEARDOWN:
+ handle_teardown_request (client, ctx);
+ break;
+ case GST_RTSP_SET_PARAMETER:
+ klass->handle_set_param_request (client, ctx);
+ break;
+ case GST_RTSP_GET_PARAMETER:
+ klass->handle_get_param_request (client, ctx);
+ break;
+ case GST_RTSP_ANNOUNCE:
+ case GST_RTSP_RECORD:
+ case GST_RTSP_REDIRECT:
+ goto not_implemented;
+ case GST_RTSP_INVALID:
+ default:
+ goto bad_request;
+ }
+
+done:
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
+ if (session)
+ g_object_unref (session);
+ if (uri)
+ gst_rtsp_url_free (uri);
+ return;
+
+ /* ERRORS */
+not_supported:
+ {
+ GST_ERROR ("client %p: version %d not supported", client, version);
+ send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
+ ctx);
+ goto done;
+ }
+bad_request:
+ {
+ GST_ERROR ("client %p: bad request", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ goto done;
+ }
+no_pool:
+ {
+ GST_ERROR ("client %p: no pool configured", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ goto done;
+ }
+session_not_found:
+ {
+ GST_ERROR ("client %p: session not found", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ goto done;
+ }
+not_authorized:
+ {
+ GST_ERROR ("client %p: not allowed", client);
+ /* error reply is already sent */
+ goto done;
+ }
+#if 0
+unsupported_requirement:
+ {
+ GST_ERROR ("client %p: Required option is not supported (%s)", client,
+ unsupported_reqs);
+ send_option_not_supported_response (client, ctx, unsupported_reqs);
+ g_free (unsupported_reqs);
+ goto done;
+ }
+#endif
+not_implemented:
+ {
+ GST_ERROR ("client %p: method %d not implemented", client, method);
+ send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
+ goto done;
+ }
+}
+
+
+static void
+handle_response (GstRTSPClient * client, GstRTSPMessage * response)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ GstRTSPSession *session = NULL;
+ GstRTSPContext sctx = { NULL }, *ctx;
+ gchar *sessid;
+
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->request = NULL;
+ ctx->uri = NULL;
+ ctx->method = GST_RTSP_INVALID;
+ ctx->response = response;
+
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (response);
+ }
+
+ GST_INFO ("client %p: received a response", client);
+
+ /* get the session if there is any */
+ res =
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
+ if (res == GST_RTSP_OK) {
+ if (priv->session_pool == NULL)
+ goto no_pool;
+
+ /* we had a session in the request, find it again */
+ if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
+ goto session_not_found;
+
+ /* we add the session to the client list of watched sessions. When a session
+ * disappears because it times out, we will be notified. If all sessions are
+ * gone, we will close the connection */
+ client_watch_session (client, session);
+ }
+
+ ctx->session = session;
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
+ 0, ctx);
+
+done:
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
+ if (session)
+ g_object_unref (session);
+ return;
+
+no_pool:
+ {
+ GST_ERROR ("client %p: no pool configured", client);
+ goto done;
+ }
+session_not_found:
+ {
+ GST_ERROR ("client %p: session not found", client);
+ goto done;
+ }
+}
+
+static void
+handle_data (GstRTSPClient * client, GstRTSPMessage * message)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ guint8 channel;
+ GList *walk;
+ guint8 *data;
+ guint size;
+ GstBuffer *buffer;
+ gboolean handled;
+
+ /* find the stream for this message */
+ res = gst_rtsp_message_parse_data (message, &channel);
+ if (res != GST_RTSP_OK)
+ return;
+
+ gst_rtsp_message_steal_body (message, &data, &size);
+
+ buffer = gst_buffer_new_wrapped (data, size);
+
+ handled = FALSE;
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *trans;
+ GstRTSPStream *stream;
+ const GstRTSPTransport *tr;
+
+ trans = walk->data;
+
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+ stream = gst_rtsp_stream_transport_get_stream (trans);
+
+ /* check for TCP transport */
+ if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+ /* dispatch to the stream based on the channel number */
+ if (tr->interleaved.min == channel) {
+ gst_rtsp_stream_recv_rtp (stream, buffer);
+ handled = TRUE;
+ break;
+ } else if (tr->interleaved.max == channel) {
+ gst_rtsp_stream_recv_rtcp (stream, buffer);
+ handled = TRUE;
+ break;
+ }
+ }
+ }
+ if (!handled)
+ gst_buffer_unref (buffer);
+}
+
+/**
+ * gst_rtsp_client_set_session_pool:
+ * @client: a #GstRTSPClient
+ * @pool: (transfer none): a #GstRTSPSessionPool
+ *
+ * Set @pool as the sessionpool for @client which it will use to find
+ * or allocate sessions. the sessionpool is usually inherited from the server
+ * that created the client but can be overridden later.
+ */
+void
+gst_rtsp_client_set_session_pool (GstRTSPClient * client,
+ GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPool *old;
+ GstRTSPClientPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ if (pool)
+ g_object_ref (pool);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->session_pool;
+ priv->session_pool = pool;
+
+ if (priv->session_removed_id) {
+ g_signal_handler_disconnect (old, priv->session_removed_id);
+ priv->session_removed_id = 0;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ /* FIXME, should remove all sessions from the old pool for this client */
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_client_get_session_pool:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
+ *
+ * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
+ */
+GstRTSPSessionPool *
+gst_rtsp_client_get_session_pool (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPSessionPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->session_pool))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_client_set_mount_points:
+ * @client: a #GstRTSPClient
+ * @mounts: (transfer none): a #GstRTSPMountPoints
+ *
+ * Set @mounts as the mount points for @client which it will use to map urls
+ * to media streams. These mount points are usually inherited from the server that
+ * created the client but can be overriden later.
+ */
+void
+gst_rtsp_client_set_mount_points (GstRTSPClient * client,
+ GstRTSPMountPoints * mounts)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPMountPoints *old;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ if (mounts)
+ g_object_ref (mounts);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->mount_points;
+ priv->mount_points = mounts;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_client_get_mount_points:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
+ *
+ * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
+ */
+GstRTSPMountPoints *
+gst_rtsp_client_get_mount_points (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPMountPoints *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->mount_points))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_client_set_auth:
+ * @client: a #GstRTSPClient
+ * @auth: (transfer none): a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @client.
+ */
+void
+gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPAuth *old;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ if (auth)
+ g_object_ref (auth);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->auth;
+ priv->auth = auth;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+
+/**
+ * gst_rtsp_client_get_auth:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @client.
+ *
+ * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
+ * usage.
+ */
+GstRTSPAuth *
+gst_rtsp_client_get_auth (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPAuth *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->auth))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_client_set_thread_pool:
+ * @client: a #GstRTSPClient
+ * @pool: (transfer none): a #GstRTSPThreadPool
+ *
+ * configure @pool to be used as the thread pool of @client.
+ */
+void
+gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
+ GstRTSPThreadPool * pool)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPThreadPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ if (pool)
+ g_object_ref (pool);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->thread_pool;
+ priv->thread_pool = pool;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_client_get_thread_pool:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPThreadPool used as the thread pool of @client.
+ *
+ * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
+ * usage.
+ */
+GstRTSPThreadPool *
+gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv;
+ GstRTSPThreadPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->thread_pool))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_client_set_connection:
+ * @client: a #GstRTSPClient
+ * @conn: (transfer full): a #GstRTSPConnection
+ *
+ * Set the #GstRTSPConnection of @client. This function takes ownership of
+ * @conn.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_client_set_connection (GstRTSPClient * client,
+ GstRTSPConnection * conn)
+{
+ GstRTSPClientPrivate *priv;
+ GSocket *read_socket;
+ GSocketAddress *address;
+ GstRTSPUrl *url;
+ GError *error = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
+ g_return_val_if_fail (conn != NULL, FALSE);
+
+ priv = client->priv;
+
+ read_socket = gst_rtsp_connection_get_read_socket (conn);
+
+ if (!(address = g_socket_get_local_address (read_socket, &error)))
+ goto no_address;
+
+ g_free (priv->server_ip);
+ /* keep the original ip that the client connected to */
+ if (G_IS_INET_SOCKET_ADDRESS (address)) {
+ GInetAddress *iaddr;
+
+ iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
+
+ /* socket might be ipv6 but adress still ipv4 */
+ priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
+ priv->server_ip = g_inet_address_to_string (iaddr);
+ g_object_unref (address);
+ } else {
+ priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
+ priv->server_ip = g_strdup ("unknown");
+ }
+
+ GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
+ priv->server_ip, priv->is_ipv6);
+
+ url = gst_rtsp_connection_get_url (conn);
+ GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
+
+ priv->connection = conn;
+
+ return TRUE;
+
+ /* ERRORS */
+no_address:
+ {
+ GST_ERROR ("could not get local address %s", error->message);
+ g_error_free (error);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_client_get_connection:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPConnection of @client.
+ *
+ * Returns: (transfer none): the #GstRTSPConnection of @client.
+ * The connection object returned remains valid until the client is freed.
+ */
+GstRTSPConnection *
+gst_rtsp_client_get_connection (GstRTSPClient * client)
+{
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ return client->priv->connection;
+}
+
+/**
+ * gst_rtsp_client_set_send_func:
+ * @client: a #GstRTSPClient
+ * @func: (scope notified): a #GstRTSPClientSendFunc
+ * @user_data: (closure): user data passed to @func
+ * @notify: (allow-none): called when @user_data is no longer in use
+ *
+ * Set @func as the callback that will be called when a new message needs to be
+ * sent to the client. @user_data is passed to @func and @notify is called when
+ * @user_data is no longer in use.
+ *
+ * By default, the client will send the messages on the #GstRTSPConnection that
+ * was configured with gst_rtsp_client_attach() was called.
+ */
+void
+gst_rtsp_client_set_send_func (GstRTSPClient * client,
+ GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
+{
+ GstRTSPClientPrivate *priv;
+ GDestroyNotify old_notify;
+ gpointer old_data;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->send_lock);
+ priv->send_func = func;
+ old_notify = priv->send_notify;
+ old_data = priv->send_data;
+ priv->send_notify = notify;
+ priv->send_data = user_data;
+ g_mutex_unlock (&priv->send_lock);
+
+ if (old_notify)
+ old_notify (old_data);
+}
+
+/**
+ * gst_rtsp_client_handle_message:
+ * @client: a #GstRTSPClient
+ * @message: (transfer none): an #GstRTSPMessage
+ *
+ * Let the client handle @message.
+ *
+ * Returns: a #GstRTSPResult.
+ */
+GstRTSPResult
+gst_rtsp_client_handle_message (GstRTSPClient * client,
+ GstRTSPMessage * message)
+{
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+
+ switch (message->type) {
+ case GST_RTSP_MESSAGE_REQUEST:
+ handle_request (client, message);
+ break;
+ case GST_RTSP_MESSAGE_RESPONSE:
+ handle_response (client, message);
+ break;
+ case GST_RTSP_MESSAGE_DATA:
+ handle_data (client, message);
+ break;
+ default:
+ break;
+ }
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_client_send_message:
+ * @client: a #GstRTSPClient
+ * @session: (allow-none) (transfer none): a #GstRTSPSession to send
+ * the message to or %NULL
+ * @message: (transfer none): The #GstRTSPMessage to send
+ *
+ * Send a message message to the remote end. @message must be a
+ * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
+ */
+GstRTSPResult
+gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPMessage * message)
+{
+ GstRTSPContext sctx = { NULL }
+ , *ctx;
+ GstRTSPClientPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
+ message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
+
+ priv = client->priv;
+
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->session = session;
+
+ send_message (client, ctx, message, FALSE);
+
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult ret;
+ GTimeVal time;
+
+ time.tv_sec = 1;
+ time.tv_usec = 0;
+
+ do {
+ /* send the response and store the seq number so we can wait until it's
+ * written to the client to close the connection */
+ ret =
+ gst_rtsp_watch_send_message (priv->watch, message,
+ close ? &priv->close_seq : NULL);
+ if (ret == GST_RTSP_OK)
+ break;
+
+ if (ret != GST_RTSP_ENOMEM)
+ goto error;
+
+ /* drop backlog */
+ if (priv->drop_backlog)
+ break;
+
+ /* queue was full, wait for more space */
+ GST_DEBUG_OBJECT (client, "waiting for backlog");
+ ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
+ GST_DEBUG_OBJECT (client, "Resend due to backlog full");
+ } while (ret != GST_RTSP_EINTR);
+
+ return ret;
+
+ /* ERRORS */
+error:
+ {
+ GST_DEBUG_OBJECT (client, "got error %d", ret);
+ return ret;
+ }
+}
+
+static GstRTSPResult
+message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
+ gpointer user_data)
+{
+ return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
+}
+
+static GstRTSPResult
+message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
+
+ if (priv->close_seq && priv->close_seq == cseq) {
+ GST_INFO ("client %p: send close message", client);
+ priv->close_seq = 0;
+ gst_rtsp_client_close (client);
+ }
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+closed (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
+ const gchar *tunnelid;
+
+ GST_INFO ("client %p: connection closed", client);
+
+ if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
+ g_mutex_lock (&tunnels_lock);
+ /* remove from tunnelids */
+ g_hash_table_remove (tunnels, tunnelid);
+ g_mutex_unlock (&tunnels_lock);
+ }
+
+ gst_rtsp_watch_set_flushing (watch, TRUE);
+ g_mutex_lock (&priv->watch_lock);
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ g_mutex_unlock (&priv->watch_lock);
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ gchar *str;
+
+ str = gst_rtsp_strresult (result);
+ GST_INFO ("client %p: received an error %s", client, str);
+ g_free (str);
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+error_full (GstRTSPWatch * watch, GstRTSPResult result,
+ GstRTSPMessage * message, guint id, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ gchar *str;
+
+ str = gst_rtsp_strresult (result);
+ GST_INFO
+ ("client %p: error when handling message %p with id %d: %s",
+ client, message, id, str);
+ g_free (str);
+
+ return GST_RTSP_OK;
+}
+
+static gboolean
+remember_tunnel (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ const gchar *tunnelid;
+
+ /* store client in the pending tunnels */
+ tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
+ if (tunnelid == NULL)
+ goto no_tunnelid;
+
+ GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
+
+ /* we can't have two clients connecting with the same tunnelid */
+ g_mutex_lock (&tunnels_lock);
+ if (g_hash_table_lookup (tunnels, tunnelid))
+ goto tunnel_existed;
+
+ g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
+ g_mutex_unlock (&tunnels_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+no_tunnelid:
+ {
+ GST_ERROR ("client %p: no tunnelid provided", client);
+ return FALSE;
+ }
+tunnel_existed:
+ {
+ g_mutex_unlock (&tunnels_lock);
+ GST_ERROR ("client %p: tunnel session %s already existed", client,
+ tunnelid);
+ return FALSE;
+ }
+}
+
+static GstRTSPResult
+tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_WARNING ("client %p: tunnel lost (connection %p)", client,
+ priv->connection);
+
+ /* ignore error, it'll only be a problem when the client does a POST again */
+ remember_tunnel (client);
+
+ return GST_RTSP_OK;
+}
+
+static gboolean
+handle_tunnel (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPClient *oclient;
+ GstRTSPClientPrivate *opriv;
+ const gchar *tunnelid;
+
+ tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
+ if (tunnelid == NULL)
+ goto no_tunnelid;
+
+ /* check for previous tunnel */
+ g_mutex_lock (&tunnels_lock);
+ oclient = g_hash_table_lookup (tunnels, tunnelid);
+
+ if (oclient == NULL) {
+ /* no previous tunnel, remember tunnel */
+ g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
+ g_mutex_unlock (&tunnels_lock);
+
+ GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
+ client, priv->connection);
+ } else {
+ /* merge both tunnels into the first client */
+ /* remove the old client from the table. ref before because removing it will
+ * remove the ref to it. */
+ g_object_ref (oclient);
+ g_hash_table_remove (tunnels, tunnelid);
+ g_mutex_unlock (&tunnels_lock);
+
+ opriv = oclient->priv;
+
+ g_mutex_lock (&opriv->watch_lock);
+ if (opriv->watch == NULL)
+ goto tunnel_closed;
+
+ GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
+ oclient, opriv->connection, priv->connection);
+
+ gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
+ gst_rtsp_watch_reset (priv->watch);
+ gst_rtsp_watch_reset (opriv->watch);
+ g_mutex_unlock (&opriv->watch_lock);
+ g_object_unref (oclient);
+
+ /* the old client owns the tunnel now, the new one will be freed */
+ g_source_destroy ((GSource *) priv->watch);
+ priv->watch = NULL;
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+no_tunnelid:
+ {
+ GST_ERROR ("client %p: no tunnelid provided", client);
+ return FALSE;
+ }
+tunnel_closed:
+ {
+ GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
+ g_mutex_unlock (&opriv->watch_lock);
+ g_object_unref (oclient);
+ return FALSE;
+ }
+}
+
+static GstRTSPStatusCode
+tunnel_get (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+ GST_INFO ("client %p: tunnel get (connection %p)", client,
+ client->priv->connection);
+
+ if (!handle_tunnel (client)) {
+ return GST_RTSP_STS_SERVICE_UNAVAILABLE;
+ }
+
+ return GST_RTSP_STS_OK;
+}
+
+static GstRTSPResult
+tunnel_post (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+ GST_INFO ("client %p: tunnel post (connection %p)", client,
+ client->priv->connection);
+
+ if (!handle_tunnel (client)) {
+ return GST_RTSP_ERROR;
+ }
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
+ GstRTSPMessage * response, gpointer user_data)
+{
+ GstRTSPClientClass *klass;
+
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+ if (klass->tunnel_http_response) {
+ klass->tunnel_http_response (client, request, response);
+ }
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPWatchFuncs watch_funcs = {
+ message_received,
+ message_sent,
+ closed,
+ error,
+ tunnel_get,
+ tunnel_post,
+ error_full,
+ tunnel_lost,
+ tunnel_http_response
+};
+
+static void
+client_watch_notify (GstRTSPClient * client)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_INFO ("client %p: watch destroyed", client);
+ priv->watch = NULL;
+ g_main_context_unref (priv->watch_context);
+ priv->watch_context = NULL;
+ /* remove all sessions and so drop the extra client ref */
+ gst_rtsp_client_session_filter (client, cleanup_session, NULL);
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
+ g_object_unref (client);
+}
+
+/**
+ * gst_rtsp_client_attach:
+ * @client: a #GstRTSPClient
+ * @context: (allow-none): a #GMainContext
+ *
+ * Attaches @client to @context. When the mainloop for @context is run, the
+ * client will be dispatched. When @context is %NULL, the default context will be
+ * used).
+ *
+ * This function should be called when the client properties and urls are fully
+ * configured and the client is ready to start.
+ *
+ * Returns: the ID (greater than 0) for the source within the GMainContext.
+ */
+guint
+gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
+{
+ GstRTSPClientPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
+ priv = client->priv;
+ g_return_val_if_fail (priv->connection != NULL, 0);
+ g_return_val_if_fail (priv->watch == NULL, 0);
+
+ /* make sure noone will free the context before the watch is destroyed */
+ priv->watch_context = g_main_context_ref (context);
+
+ /* create watch for the connection and attach */
+ priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
+ g_object_ref (client), (GDestroyNotify) client_watch_notify);
+ gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
+ (GDestroyNotify) gst_rtsp_watch_unref);
+
+ /* FIXME make this configurable. We don't want to do this yet because it will
+ * be superceeded by a cache object later */
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100);
+
+ GST_INFO ("client %p: attaching to context %p", client, context);
+ res = gst_rtsp_watch_attach (priv->watch, context);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_client_session_filter:
+ * @client: a #GstRTSPClient
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: user data passed to @func
+ *
+ * Call @func for each session managed by @client. The result value of @func
+ * determines what happens to the session. @func will be called with @client
+ * locked so no further actions on @client can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
+ * @client.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
+ *
+ * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
+ * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_client_session_filter (GstRTSPClient * client,
+ GstRTSPClientSessionFilterFunc func, gpointer user_data)
+{
+ GstRTSPClientPrivate *priv;
+ GList *result, *walk, *next;
+ GHashTable *visited;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ g_mutex_lock (&priv->lock);
+restart:
+ cookie = priv->sessions_cookie;
+ for (walk = priv->sessions; walk; walk = next) {
+ GstRTSPSession *sess = walk->data;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ next = g_list_next (walk);
+
+ if (func) {
+ /* only visit each session once */
+ if (g_hash_table_contains (visited, sess))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (sess));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (client, sess, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->sessions_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ /* stop watching the session and pretend it went away, if the list was
+ * changed, we can't use the current list position, try to see if we
+ * still have the session */
+ client_unwatch_session (client, sess, changed ? NULL : walk);
+ cookie = priv->sessions_cookie;
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (sess));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspconnection.h>
+
+#ifndef __GST_RTSP_CLIENT_H__
+#define __GST_RTSP_CLIENT_H__
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPClient GstRTSPClient;
+typedef struct _GstRTSPClientClass GstRTSPClientClass;
+typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate;
+
+#include "rtsp-context.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-sdp.h"
+#include "rtsp-auth.h"
+
+#define GST_TYPE_RTSP_CLIENT (gst_rtsp_client_get_type ())
+#define GST_IS_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CLIENT))
+#define GST_IS_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CLIENT))
+#define GST_RTSP_CLIENT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
+#define GST_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClient))
+#define GST_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
+#define GST_RTSP_CLIENT_CAST(obj) ((GstRTSPClient*)(obj))
+#define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass))
+
+/**
+ * GstRTSPClientSendFunc:
+ * @client: a #GstRTSPClient
+ * @message: a #GstRTSPMessage
+ * @close: close the connection
+ * @user_data: user data when registering the callback
+ *
+ * This callback is called when @client wants to send @message. When @close is
+ * %TRUE, the connection should be closed when the message has been sent.
+ *
+ * Returns: %TRUE on success.
+ */
+typedef gboolean (*GstRTSPClientSendFunc) (GstRTSPClient *client,
+ GstRTSPMessage *message,
+ gboolean close,
+ gpointer user_data);
+
+/**
+ * GstRTSPClient:
+ *
+ * The client object represents the connection and its state with a client.
+ */
+struct _GstRTSPClient {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPClientPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPClientClass:
+ * @create_sdp: called when the SDP needs to be created for media.
+ * @configure_client_media: called when the stream in media needs to be configured.
+ * The default implementation will configure the blocksize on the payloader when
+ * spcified in the request headers.
+ * @configure_client_transport: called when the client transport needs to be
+ * configured.
+ * @params_set: set parameters. This function should also initialize the
+ * RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
+ * @params_get: get parameters. This function should also initialize the
+ * RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
+ * @tunnel_http_response: called when a response to the GET request is about to
+ * be sent for a tunneled connection. The response can be modified. Since 1.4
+ *
+ * The client class structure.
+ */
+struct _GstRTSPClientClass {
+ GObjectClass parent_class;
+
+ GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
+ gboolean (*configure_client_media) (GstRTSPClient * client,
+ GstRTSPMedia * media, GstRTSPStream * stream,
+ GstRTSPContext * ctx);
+ gboolean (*configure_client_transport) (GstRTSPClient * client,
+ GstRTSPContext * ctx,
+ GstRTSPTransport * ct);
+ GstRTSPResult (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx);
+ GstRTSPResult (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
+ gchar * (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri);
+ gboolean (*handle_options_request) (GstRTSPClient * client, GstRTSPContext * ctx);
+ gboolean (*handle_set_param_request) (GstRTSPClient * client, GstRTSPContext * ctx);
+ gboolean (*handle_get_param_request) (GstRTSPClient * client, GstRTSPContext * ctx);
+
+ /* signals */
+ void (*closed) (GstRTSPClient *client);
+ void (*new_session) (GstRTSPClient *client, GstRTSPSession *session);
+ void (*options_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*describe_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*setup_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*play_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*pause_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
+ void (*handle_response) (GstRTSPClient *client, GstRTSPContext *ctx);
+
+ void (*tunnel_http_response) (GstRTSPClient * client, GstRTSPMessage * request,
+ GstRTSPMessage * response);
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE-1];
+};
+
+GType gst_rtsp_client_get_type (void);
+
+GstRTSPClient * gst_rtsp_client_new (void);
+
+void gst_rtsp_client_set_session_pool (GstRTSPClient *client,
+ GstRTSPSessionPool *pool);
+GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client);
+
+void gst_rtsp_client_set_mount_points (GstRTSPClient *client,
+ GstRTSPMountPoints *mounts);
+GstRTSPMountPoints * gst_rtsp_client_get_mount_points (GstRTSPClient *client);
+
+void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth);
+GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client);
+
+void gst_rtsp_client_set_thread_pool (GstRTSPClient *client, GstRTSPThreadPool *pool);
+GstRTSPThreadPool * gst_rtsp_client_get_thread_pool (GstRTSPClient *client);
+
+gboolean gst_rtsp_client_set_connection (GstRTSPClient *client, GstRTSPConnection *conn);
+GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient *client);
+
+guint gst_rtsp_client_attach (GstRTSPClient *client,
+ GMainContext *context);
+void gst_rtsp_client_close (GstRTSPClient * client);
+
+void gst_rtsp_client_set_send_func (GstRTSPClient *client,
+ GstRTSPClientSendFunc func,
+ gpointer user_data,
+ GDestroyNotify notify);
+
+GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client,
+ GstRTSPMessage *message);
+GstRTSPResult gst_rtsp_client_send_message (GstRTSPClient * client,
+ GstRTSPSession *session,
+ GstRTSPMessage *message);
+/**
+ * GstRTSPClientSessionFilterFunc:
+ * @client: a #GstRTSPClient object
+ * @sess: a #GstRTSPSession in @client
+ * @user_data: user data that has been given to gst_rtsp_client_session_filter()
+ *
+ * This function will be called by the gst_rtsp_client_session_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed
+ * from @client.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in
+ * @client.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of
+ * gst_rtsp_client_session_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPClientSessionFilterFunc) (GstRTSPClient *client,
+ GstRTSPSession *sess,
+ gpointer user_data);
+
+GList * gst_rtsp_client_session_filter (GstRTSPClient *client,
+ GstRTSPClientSessionFilterFunc func,
+ gpointer user_data);
+
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_CLIENT_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-context
+ * @short_description: A client request context
+ * @see_also: #GstRTSPServer, #GstRTSPClient
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include "rtsp-context.h"
+
+G_DEFINE_POINTER_TYPE (GstRTSPContext, gst_rtsp_context);
+
+static GPrivate current_context;
+
+/**
+ * gst_rtsp_context_get_current:
+ *
+ * Get the current #GstRTSPContext. This object is retrieved from the
+ * current thread that is handling the request for a client.
+ *
+ * Returns: a #GstRTSPContext
+ */
+GstRTSPContext *
+gst_rtsp_context_get_current (void)
+{
+ GSList *l;
+
+ l = g_private_get (¤t_context);
+ if (l == NULL)
+ return NULL;
+
+ return (GstRTSPContext *) (l->data);
+
+}
+
+/**
+ * gst_rtsp_context_push_current:
+ * @ctx: a ##GstRTSPContext
+ *
+ * Pushes @ctx onto the context stack. The current
+ * context can then be received using gst_rtsp_context_get_current().
+ **/
+void
+gst_rtsp_context_push_current (GstRTSPContext * ctx)
+{
+ GSList *l;
+
+ g_return_if_fail (ctx != NULL);
+
+ l = g_private_get (¤t_context);
+ l = g_slist_prepend (l, ctx);
+ g_private_set (¤t_context, l);
+}
+
+/**
+ * gst_rtsp_context_pop_current:
+ * @ctx: a #GstRTSPContext
+ *
+ * Pops @ctx off the context stack (verifying that @ctx
+ * is on the top of the stack).
+ **/
+void
+gst_rtsp_context_pop_current (GstRTSPContext * ctx)
+{
+ GSList *l;
+
+ l = g_private_get (¤t_context);
+
+ g_return_if_fail (l != NULL);
+ g_return_if_fail (l->data == ctx);
+
+ l = g_slist_delete_link (l, l);
+ g_private_set (¤t_context, l);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspconnection.h>
+
+#ifndef __GST_RTSP_CONTEXT_H__
+#define __GST_RTSP_CONTEXT_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_CONTEXT (gst_rtsp_context_get_type ())
+
+typedef struct _GstRTSPContext GstRTSPContext;
+
+#include "rtsp-server.h"
+#include "rtsp-media.h"
+#include "rtsp-media-factory.h"
+#include "rtsp-session-media.h"
+#include "rtsp-auth.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-token.h"
+
+/**
+ * GstRTSPContext:
+ * @server: the server
+ * @conn: the connection
+ * @client: the client
+ * @request: the complete request
+ * @uri: the complete url parsed from @request
+ * @method: the parsed method of @uri
+ * @auth: the current auth object or %NULL
+ * @token: authorisation token
+ * @session: the session, can be %NULL
+ * @sessmedia: the session media for the url can be %NULL
+ * @factory: the media factory for the url, can be %NULL
+ * @media: the media for the url can be %NULL
+ * @stream: the stream for the url can be %NULL
+ * @response: the response
+ *
+ * Information passed around containing the context of a request.
+ */
+struct _GstRTSPContext {
+ GstRTSPServer *server;
+ GstRTSPConnection *conn;
+ GstRTSPClient *client;
+ GstRTSPMessage *request;
+ GstRTSPUrl *uri;
+ GstRTSPMethod method;
+ GstRTSPAuth *auth;
+ GstRTSPToken *token;
+ GstRTSPSession *session;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPStream *stream;
+ GstRTSPMessage *response;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_context_get_type (void);
+
+GstRTSPContext * gst_rtsp_context_get_current (void);
+void gst_rtsp_context_push_current (GstRTSPContext * ctx);
+void gst_rtsp_context_pop_current (GstRTSPContext * ctx);
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_CONTEXT_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media-factory-uri
+ * @short_description: A factory for URI sources
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPMedia
+ *
+ * This specialized #GstRTSPMediaFactory constructs media pipelines from a URI,
+ * given with gst_rtsp_media_factory_uri_set_uri().
+ *
+ * It will automatically demux and payload the different streams found in the
+ * media at URL.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include <string.h>
+
+#include "rtsp-media-factory-uri.h"
+
+#define GST_RTSP_MEDIA_FACTORY_URI_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURIPrivate))
+
+struct _GstRTSPMediaFactoryURIPrivate
+{
+ GMutex lock;
+ gchar *uri; /* protected by lock */
+ gboolean use_gstpay;
+
+ GstCaps *raw_vcaps;
+ GstCaps *raw_acaps;
+ GList *demuxers;
+ GList *payloaders;
+ GList *decoders;
+};
+
+#define DEFAULT_URI NULL
+#define DEFAULT_USE_GSTPAY FALSE
+
+enum
+{
+ PROP_0,
+ PROP_URI,
+ PROP_USE_GSTPAY,
+ PROP_LAST
+};
+
+
+#define RAW_VIDEO_CAPS \
+ "video/x-raw"
+
+#define RAW_AUDIO_CAPS \
+ "audio/x-raw"
+
+static GstStaticCaps raw_video_caps = GST_STATIC_CAPS (RAW_VIDEO_CAPS);
+static GstStaticCaps raw_audio_caps = GST_STATIC_CAPS (RAW_AUDIO_CAPS);
+
+typedef struct
+{
+ GstRTSPMediaFactoryURI *factory;
+ guint pt;
+} FactoryData;
+
+static void
+free_data (FactoryData * data)
+{
+ g_object_unref (data->factory);
+ g_free (data);
+}
+
+static const gchar *factory_key = "GstRTSPMediaFactoryURI";
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_factory_uri_debug);
+#define GST_CAT_DEFAULT rtsp_media_factory_uri_debug
+
+static void gst_rtsp_media_factory_uri_get_property (GObject * object,
+ guint propid, GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_uri_set_property (GObject * object,
+ guint propid, const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_uri_finalize (GObject * obj);
+
+static GstElement *rtsp_media_factory_uri_create_element (GstRTSPMediaFactory *
+ factory, const GstRTSPUrl * url);
+
+G_DEFINE_TYPE (GstRTSPMediaFactoryURI, gst_rtsp_media_factory_uri,
+ GST_TYPE_RTSP_MEDIA_FACTORY);
+
+static void
+gst_rtsp_media_factory_uri_class_init (GstRTSPMediaFactoryURIClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstRTSPMediaFactoryClass *mediafactory_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPMediaFactoryURIPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ mediafactory_class = GST_RTSP_MEDIA_FACTORY_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_media_factory_uri_get_property;
+ gobject_class->set_property = gst_rtsp_media_factory_uri_set_property;
+ gobject_class->finalize = gst_rtsp_media_factory_uri_finalize;
+
+ /**
+ * GstRTSPMediaFactoryURI::uri:
+ *
+ * The uri of the resource that will be served by this factory.
+ */
+ g_object_class_install_property (gobject_class, PROP_URI,
+ g_param_spec_string ("uri", "URI",
+ "The URI of the resource to stream", DEFAULT_URI,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPMediaFactoryURI::use-gstpay:
+ *
+ * Allow the usage of gstpay in order to avoid decoding of compressed formats
+ * without a payloader.
+ */
+ g_object_class_install_property (gobject_class, PROP_USE_GSTPAY,
+ g_param_spec_boolean ("use-gstpay", "Use gstpay",
+ "Use the gstpay payloader to avoid decoding", DEFAULT_USE_GSTPAY,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ mediafactory_class->create_element = rtsp_media_factory_uri_create_element;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_media_factory_uri_debug, "rtspmediafactoryuri",
+ 0, "GstRTSPMediaFactoryUri");
+}
+
+typedef struct
+{
+ GList *demux;
+ GList *payload;
+ GList *decode;
+} FilterData;
+
+static gboolean
+payloader_filter (GstPluginFeature * feature, FilterData * data)
+{
+ const gchar *klass;
+ GstElementFactory *fact;
+ GList **list = NULL;
+
+ /* we only care about element factories */
+ if (G_UNLIKELY (!GST_IS_ELEMENT_FACTORY (feature)))
+ return FALSE;
+
+ if (gst_plugin_feature_get_rank (feature) < GST_RANK_MARGINAL)
+ return FALSE;
+
+ fact = GST_ELEMENT_FACTORY_CAST (feature);
+
+ klass = gst_element_factory_get_metadata (fact, GST_ELEMENT_METADATA_KLASS);
+
+ if (strstr (klass, "Decoder"))
+ list = &data->decode;
+ else if (strstr (klass, "Demux"))
+ list = &data->demux;
+ else if (strstr (klass, "Parser") && strstr (klass, "Codec"))
+ list = &data->demux;
+ else if (strstr (klass, "Payloader") && strstr (klass, "RTP"))
+ list = &data->payload;
+
+ if (list) {
+ GST_DEBUG ("adding %s", GST_OBJECT_NAME (fact));
+ *list = g_list_prepend (*list, gst_object_ref (fact));
+ }
+
+ return FALSE;
+}
+
+static void
+gst_rtsp_media_factory_uri_init (GstRTSPMediaFactoryURI * factory)
+{
+ GstRTSPMediaFactoryURIPrivate *priv =
+ GST_RTSP_MEDIA_FACTORY_URI_GET_PRIVATE (factory);
+ FilterData data = { NULL, NULL, NULL };
+
+ GST_DEBUG_OBJECT (factory, "new");
+
+ factory->priv = priv;
+
+ priv->uri = g_strdup (DEFAULT_URI);
+ priv->use_gstpay = DEFAULT_USE_GSTPAY;
+ g_mutex_init (&priv->lock);
+
+ /* get the feature list using the filter */
+ gst_registry_feature_filter (gst_registry_get (), (GstPluginFeatureFilter)
+ payloader_filter, FALSE, &data);
+ /* sort */
+ priv->demuxers =
+ g_list_sort (data.demux, gst_plugin_feature_rank_compare_func);
+ priv->payloaders =
+ g_list_sort (data.payload, gst_plugin_feature_rank_compare_func);
+ priv->decoders =
+ g_list_sort (data.decode, gst_plugin_feature_rank_compare_func);
+
+ priv->raw_vcaps = gst_static_caps_get (&raw_video_caps);
+ priv->raw_acaps = gst_static_caps_get (&raw_audio_caps);
+}
+
+static void
+gst_rtsp_media_factory_uri_finalize (GObject * obj)
+{
+ GstRTSPMediaFactoryURI *factory = GST_RTSP_MEDIA_FACTORY_URI (obj);
+ GstRTSPMediaFactoryURIPrivate *priv = factory->priv;
+
+ GST_DEBUG_OBJECT (factory, "finalize");
+
+ g_free (priv->uri);
+ gst_plugin_feature_list_free (priv->demuxers);
+ gst_plugin_feature_list_free (priv->payloaders);
+ gst_plugin_feature_list_free (priv->decoders);
+ gst_caps_unref (priv->raw_vcaps);
+ gst_caps_unref (priv->raw_acaps);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_media_factory_uri_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_media_factory_uri_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMediaFactoryURI *factory = GST_RTSP_MEDIA_FACTORY_URI (object);
+ GstRTSPMediaFactoryURIPrivate *priv = factory->priv;
+
+ switch (propid) {
+ case PROP_URI:
+ g_value_take_string (value, gst_rtsp_media_factory_uri_get_uri (factory));
+ break;
+ case PROP_USE_GSTPAY:
+ g_value_set_boolean (value, priv->use_gstpay);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_media_factory_uri_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMediaFactoryURI *factory = GST_RTSP_MEDIA_FACTORY_URI (object);
+ GstRTSPMediaFactoryURIPrivate *priv = factory->priv;
+
+ switch (propid) {
+ case PROP_URI:
+ gst_rtsp_media_factory_uri_set_uri (factory, g_value_get_string (value));
+ break;
+ case PROP_USE_GSTPAY:
+ priv->use_gstpay = g_value_get_boolean (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_media_factory_uri_new:
+ *
+ * Create a new #GstRTSPMediaFactoryURI instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPMediaFactoryURI object.
+ */
+GstRTSPMediaFactoryURI *
+gst_rtsp_media_factory_uri_new (void)
+{
+ GstRTSPMediaFactoryURI *result;
+
+ result = g_object_new (GST_TYPE_RTSP_MEDIA_FACTORY_URI, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_uri_set_uri:
+ * @factory: a #GstRTSPMediaFactory
+ * @uri: the uri the stream
+ *
+ * Set the URI of the resource that will be streamed by this factory.
+ */
+void
+gst_rtsp_media_factory_uri_set_uri (GstRTSPMediaFactoryURI * factory,
+ const gchar * uri)
+{
+ GstRTSPMediaFactoryURIPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY_URI (factory));
+ g_return_if_fail (uri != NULL);
+
+ priv = factory->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_free (priv->uri);
+ priv->uri = g_strdup (uri);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_factory_uri_get_uri:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the URI that will provide media for this factory.
+ *
+ * Returns: (transfer full): the configured URI. g_free() after usage.
+ */
+gchar *
+gst_rtsp_media_factory_uri_get_uri (GstRTSPMediaFactoryURI * factory)
+{
+ GstRTSPMediaFactoryURIPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY_URI (factory), NULL);
+
+ priv = factory->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = g_strdup (priv->uri);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+static GstElementFactory *
+find_payloader (GstRTSPMediaFactoryURI * urifact, GstCaps * caps)
+{
+ GstRTSPMediaFactoryURIPrivate *priv = urifact->priv;
+ GList *list;
+ GstElementFactory *factory = NULL;
+ gboolean autoplug_more = FALSE;
+
+ /* first find a demuxer that can link */
+ list = gst_element_factory_list_filter (priv->demuxers, caps,
+ GST_PAD_SINK, FALSE);
+
+ if (list) {
+ GstStructure *structure = gst_caps_get_structure (caps, 0);
+ gboolean parsed = FALSE;
+ gint mpegversion = 0;
+
+ if (!gst_structure_get_boolean (structure, "parsed", &parsed) &&
+ gst_structure_has_name (structure, "audio/mpeg") &&
+ gst_structure_get_int (structure, "mpegversion", &mpegversion) &&
+ (mpegversion == 2 || mpegversion == 4)) {
+ /* for AAC it's framed=true instead of parsed=true */
+ gst_structure_get_boolean (structure, "framed", &parsed);
+ }
+
+ /* Avoid plugging parsers in a loop. This is not 100% correct, as some
+ * parsers don't set parsed=true in caps. We should do something like
+ * decodebin does and track decode chains and elements plugged in those
+ * chains...
+ */
+ if (parsed) {
+ GList *walk;
+ const gchar *klass;
+
+ for (walk = list; walk; walk = walk->next) {
+ factory = GST_ELEMENT_FACTORY (walk->data);
+ klass = gst_element_factory_get_metadata (factory,
+ GST_ELEMENT_METADATA_KLASS);
+ if (strstr (klass, "Parser"))
+ /* caps have parsed=true, so skip this parser to avoid loops */
+ continue;
+
+ autoplug_more = TRUE;
+ break;
+ }
+ } else {
+ /* caps don't have parsed=true set and we have a demuxer/parser */
+ autoplug_more = TRUE;
+ }
+
+ gst_plugin_feature_list_free (list);
+ }
+
+ if (autoplug_more)
+ /* we have a demuxer, try that one first */
+ return NULL;
+
+ /* no demuxer try a depayloader */
+ list = gst_element_factory_list_filter (priv->payloaders, caps,
+ GST_PAD_SINK, FALSE);
+
+ if (list == NULL) {
+ if (priv->use_gstpay) {
+ /* no depayloader or parser/demuxer, use gstpay when allowed */
+ factory = gst_element_factory_find ("rtpgstpay");
+ } else {
+ /* no depayloader, try a decoder, we'll get to a payloader for a decoded
+ * video or audio format, worst case. */
+ list = gst_element_factory_list_filter (priv->decoders, caps,
+ GST_PAD_SINK, FALSE);
+
+ if (list != NULL) {
+ /* we have a decoder, try that one first */
+ gst_plugin_feature_list_free (list);
+ return NULL;
+ }
+ }
+ }
+
+ if (list != NULL) {
+ factory = GST_ELEMENT_FACTORY_CAST (list->data);
+ g_object_ref (factory);
+ gst_plugin_feature_list_free (list);
+ }
+ return factory;
+}
+
+static gboolean
+autoplug_continue_cb (GstElement * uribin, GstPad * pad, GstCaps * caps,
+ GstElement * element)
+{
+ FactoryData *data;
+ GstElementFactory *factory;
+
+ GST_DEBUG ("found pad %s:%s of caps %" GST_PTR_FORMAT,
+ GST_DEBUG_PAD_NAME (pad), caps);
+
+ data = g_object_get_data (G_OBJECT (element), factory_key);
+
+ if (!(factory = find_payloader (data->factory, caps)))
+ goto no_factory;
+
+ /* we found a payloader, stop autoplugging so we can plug the
+ * payloader. */
+ GST_DEBUG ("found factory %s",
+ gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (factory)));
+ gst_object_unref (factory);
+
+ return FALSE;
+
+ /* ERRORS */
+no_factory:
+ {
+ /* no payloader, continue autoplugging */
+ GST_DEBUG ("no payloader found");
+ return TRUE;
+ }
+}
+
+static void
+pad_added_cb (GstElement * uribin, GstPad * pad, GstElement * element)
+{
+ GstRTSPMediaFactoryURI *urifact;
+ GstRTSPMediaFactoryURIPrivate *priv;
+ FactoryData *data;
+ GstElementFactory *factory;
+ GstElement *payloader;
+ GstCaps *caps;
+ GstPad *sinkpad, *srcpad, *ghostpad;
+ GstElement *convert;
+ gchar *padname;
+
+ GST_DEBUG ("added pad %s:%s", GST_DEBUG_PAD_NAME (pad));
+
+ /* link the element now and expose the pad */
+ data = g_object_get_data (G_OBJECT (element), factory_key);
+ urifact = data->factory;
+ priv = urifact->priv;
+
+ /* ref to make refcounting easier later */
+ gst_object_ref (pad);
+ padname = gst_pad_get_name (pad);
+
+ /* get pad caps first, then call get_caps, then fail */
+ if ((caps = gst_pad_get_current_caps (pad)) == NULL)
+ if ((caps = gst_pad_query_caps (pad, NULL)) == NULL)
+ goto no_caps;
+
+ /* check for raw caps */
+ if (gst_caps_can_intersect (caps, priv->raw_vcaps)) {
+ /* we have raw video caps, insert converter */
+ convert = gst_element_factory_make ("videoconvert", NULL);
+ } else if (gst_caps_can_intersect (caps, priv->raw_acaps)) {
+ /* we have raw audio caps, insert converter */
+ convert = gst_element_factory_make ("audioconvert", NULL);
+ } else {
+ convert = NULL;
+ }
+
+ if (convert) {
+ gst_bin_add (GST_BIN_CAST (element), convert);
+ gst_element_set_state (convert, GST_STATE_PLAYING);
+
+ sinkpad = gst_element_get_static_pad (convert, "sink");
+ gst_pad_link (pad, sinkpad);
+ gst_object_unref (sinkpad);
+
+ /* unref old pad, we reffed before */
+ gst_object_unref (pad);
+
+ /* continue with new pad and caps */
+ pad = gst_element_get_static_pad (convert, "src");
+ if ((caps = gst_pad_get_current_caps (pad)) == NULL)
+ if ((caps = gst_pad_query_caps (pad, NULL)) == NULL)
+ goto no_caps;
+ }
+
+ if (!(factory = find_payloader (urifact, caps)))
+ goto no_factory;
+
+ gst_caps_unref (caps);
+
+ /* we have a payloader now */
+ GST_DEBUG ("found payloader factory %s",
+ gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (factory)));
+
+ payloader = gst_element_factory_create (factory, NULL);
+ if (payloader == NULL)
+ goto no_payloader;
+
+ g_object_set (payloader, "pt", data->pt, NULL);
+ data->pt++;
+
+ if (g_object_class_find_property (G_OBJECT_GET_CLASS (payloader),
+ "buffer-list"))
+ g_object_set (payloader, "buffer-list", TRUE, NULL);
+
+ /* add the payloader to the pipeline */
+ gst_bin_add (GST_BIN_CAST (element), payloader);
+ gst_element_set_state (payloader, GST_STATE_PLAYING);
+
+ /* link the pad to the sinkpad of the payloader */
+ sinkpad = gst_element_get_static_pad (payloader, "sink");
+ gst_pad_link (pad, sinkpad);
+ gst_object_unref (sinkpad);
+ gst_object_unref (pad);
+
+ /* now expose the srcpad of the payloader as a ghostpad with the same name
+ * as the uridecodebin pad name. */
+ srcpad = gst_element_get_static_pad (payloader, "src");
+ ghostpad = gst_ghost_pad_new (padname, srcpad);
+ gst_object_unref (srcpad);
+ g_free (padname);
+
+ gst_pad_set_active (ghostpad, TRUE);
+ gst_element_add_pad (element, ghostpad);
+
+ return;
+
+ /* ERRORS */
+no_caps:
+ {
+ GST_WARNING ("could not get caps from pad");
+ g_free (padname);
+ gst_object_unref (pad);
+ return;
+ }
+no_factory:
+ {
+ GST_DEBUG ("no payloader found");
+ g_free (padname);
+ gst_caps_unref (caps);
+ gst_object_unref (pad);
+ return;
+ }
+no_payloader:
+ {
+ GST_ERROR ("could not create payloader from factory");
+ g_free (padname);
+ gst_caps_unref (caps);
+ gst_object_unref (pad);
+ return;
+ }
+}
+
+static void
+no_more_pads_cb (GstElement * uribin, GstElement * element)
+{
+ GST_DEBUG ("no-more-pads");
+ gst_element_no_more_pads (element);
+}
+
+static GstElement *
+rtsp_media_factory_uri_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMediaFactoryURIPrivate *priv;
+ GstElement *topbin, *element, *uribin;
+ GstRTSPMediaFactoryURI *urifact;
+ FactoryData *data;
+
+ urifact = GST_RTSP_MEDIA_FACTORY_URI_CAST (factory);
+ priv = urifact->priv;
+
+ GST_LOG ("creating element");
+
+ topbin = gst_bin_new ("GstRTSPMediaFactoryURI");
+ g_assert (topbin != NULL);
+
+ /* our bin will dynamically expose payloaded pads */
+ element = gst_bin_new ("dynpay0");
+ g_assert (element != NULL);
+
+ uribin = gst_element_factory_make ("uridecodebin", "uribin");
+ if (uribin == NULL)
+ goto no_uridecodebin;
+
+ g_object_set (uribin, "uri", priv->uri, NULL);
+
+ /* keep factory data around */
+ data = g_new0 (FactoryData, 1);
+ data->factory = g_object_ref (factory);
+ data->pt = 96;
+
+ g_object_set_data_full (G_OBJECT (element), factory_key,
+ data, (GDestroyNotify) free_data);
+
+ /* connect to the signals */
+ g_signal_connect (uribin, "autoplug-continue",
+ (GCallback) autoplug_continue_cb, element);
+ g_signal_connect (uribin, "pad-added", (GCallback) pad_added_cb, element);
+ g_signal_connect (uribin, "no-more-pads", (GCallback) no_more_pads_cb,
+ element);
+
+ gst_bin_add (GST_BIN_CAST (element), uribin);
+ gst_bin_add (GST_BIN_CAST (topbin), element);
+
+ return topbin;
+
+no_uridecodebin:
+ {
+ g_critical ("can't create uridecodebin element");
+ gst_object_unref (element);
+ return NULL;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include "rtsp-media-factory.h"
+
+#ifndef __GST_RTSP_MEDIA_FACTORY_URI_H__
+#define __GST_RTSP_MEDIA_FACTORY_URI_H__
+
+G_BEGIN_DECLS
+
+/* types for the media factory */
+#define GST_TYPE_RTSP_MEDIA_FACTORY_URI (gst_rtsp_media_factory_uri_get_type ())
+#define GST_IS_RTSP_MEDIA_FACTORY_URI(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI))
+#define GST_IS_RTSP_MEDIA_FACTORY_URI_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA_FACTORY_URI))
+#define GST_RTSP_MEDIA_FACTORY_URI_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURIClass))
+#define GST_RTSP_MEDIA_FACTORY_URI(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURI))
+#define GST_RTSP_MEDIA_FACTORY_URI_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURIClass))
+#define GST_RTSP_MEDIA_FACTORY_URI_CAST(obj) ((GstRTSPMediaFactoryURI*)(obj))
+#define GST_RTSP_MEDIA_FACTORY_URI_CLASS_CAST(klass) ((GstRTSPMediaFactoryURIClass*)(klass))
+
+typedef struct _GstRTSPMediaFactoryURI GstRTSPMediaFactoryURI;
+typedef struct _GstRTSPMediaFactoryURIClass GstRTSPMediaFactoryURIClass;
+typedef struct _GstRTSPMediaFactoryURIPrivate GstRTSPMediaFactoryURIPrivate;
+
+/**
+ * GstRTSPMediaFactoryURI:
+ *
+ * A media factory that creates a pipeline to play and uri.
+ */
+struct _GstRTSPMediaFactoryURI {
+ GstRTSPMediaFactory parent;
+
+ /*< private >*/
+ GstRTSPMediaFactoryURIPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaFactoryURIClass:
+ *
+ * The #GstRTSPMediaFactoryURI class structure.
+ */
+struct _GstRTSPMediaFactoryURIClass {
+ GstRTSPMediaFactoryClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_media_factory_uri_get_type (void);
+
+/* creating the factory */
+GstRTSPMediaFactoryURI * gst_rtsp_media_factory_uri_new (void);
+
+/* configuring the factory */
+void gst_rtsp_media_factory_uri_set_uri (GstRTSPMediaFactoryURI *factory,
+ const gchar *uri);
+gchar * gst_rtsp_media_factory_uri_get_uri (GstRTSPMediaFactoryURI *factory);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MEDIA_FACTORY_URI_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media-factory
+ * @short_description: A factory for media pipelines
+ * @see_also: #GstRTSPMountPoints, #GstRTSPMedia
+ *
+ * The #GstRTSPMediaFactoryWFD is responsible for creating or recycling
+ * #GstRTSPMedia objects based on the passed URL.
+ *
+ * The default implementation of the object can create #GstRTSPMedia objects
+ * containing a pipeline created from a launch description set with
+ * gst_rtsp_media_factory_wfd_set_launch().
+ *
+ * Media from a factory can be shared by setting the shared flag with
+ * gst_rtsp_media_factory_wfd_set_shared(). When a factory is shared,
+ * gst_rtsp_media_factory_wfd_construct() will return the same #GstRTSPMedia when
+ * the url matches.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include <stdio.h>
+#include "rtsp-media-factory-wfd.h"
+#include "gstwfdmessage.h"
+
+#define GST_RTSP_MEDIA_FACTORY_WFD_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_WFD, GstRTSPMediaFactoryWFDPrivate))
+
+#define GST_RTSP_MEDIA_FACTORY_WFD_GET_LOCK(f) (&(GST_RTSP_MEDIA_FACTORY_WFD_CAST(f)->priv->lock))
+#define GST_RTSP_MEDIA_FACTORY_WFD_LOCK(f) (g_mutex_lock(GST_RTSP_MEDIA_FACTORY_WFD_GET_LOCK(f)))
+#define GST_RTSP_MEDIA_FACTORY_WFD_UNLOCK(f) (g_mutex_unlock(GST_RTSP_MEDIA_FACTORY_WFD_GET_LOCK(f)))
+
+struct _GstRTSPMediaFactoryWFDPrivate
+{
+ GMutex lock;
+ GstRTSPPermissions *permissions;
+ gchar *launch;
+ gboolean shared;
+ GstRTSPLowerTrans protocols;
+ guint buffer_size;
+ guint mtu_size;
+
+ guint8 videosrc_type;
+ guint8 video_codec;
+ gchar *video_encoder;
+ guint video_bitrate;
+ guint video_width;
+ guint video_height;
+ guint video_framerate;
+ guint video_enc_skip_inbuf_value;
+ GstElement *video_queue;
+
+ gchar *audio_device;
+ gchar *audio_encoder_aac;
+ gchar *audio_encoder_ac3;
+ guint8 audio_codec;
+ guint64 audio_latency_time;
+ guint64 audio_buffer_time;
+ gboolean audio_do_timestamp;
+ guint8 audio_channels;
+ guint8 audio_freq;
+ guint8 audio_bitrate;
+ GstElement *audio_queue;
+
+ guint64 video_resolution_supported;
+
+ gboolean dump_ts;
+};
+
+#define DEFAULT_LAUNCH NULL
+#define DEFAULT_SHARED FALSE
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+ GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_BUFFER_SIZE 0x80000
+
+enum
+{
+ PROP_0,
+ PROP_LAUNCH,
+ PROP_SHARED,
+ PROP_SUSPEND_MODE,
+ PROP_EOS_SHUTDOWN,
+ PROP_PROTOCOLS,
+ PROP_BUFFER_SIZE,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_MEDIA_CONSTRUCTED,
+ SIGNAL_MEDIA_CONFIGURE,
+ SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_wfd_debug);
+#define GST_CAT_DEFAULT rtsp_media_wfd_debug
+
+static void gst_rtsp_media_factory_wfd_get_property (GObject * object,
+ guint propid, GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_wfd_set_property (GObject * object,
+ guint propid, const GValue * value, GParamSpec * pspec);
+
+static void gst_rtsp_media_factory_wfd_finalize (GObject * obj);
+
+
+static GstElement *rtsp_media_factory_wfd_create_element (GstRTSPMediaFactory *
+ factory, const GstRTSPUrl * url);
+static GstRTSPMedia *rtsp_media_factory_wfd_construct (GstRTSPMediaFactory *
+ factory, const GstRTSPUrl * url);
+
+G_DEFINE_TYPE (GstRTSPMediaFactoryWFD, gst_rtsp_media_factory_wfd,
+ GST_TYPE_RTSP_MEDIA_FACTORY);
+
+static void
+gst_rtsp_media_factory_wfd_class_init (GstRTSPMediaFactoryWFDClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstRTSPMediaFactoryClass *factory_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPMediaFactoryWFDPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ factory_class = GST_RTSP_MEDIA_FACTORY_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_media_factory_wfd_get_property;
+ gobject_class->set_property = gst_rtsp_media_factory_wfd_set_property;
+ gobject_class->finalize = gst_rtsp_media_factory_wfd_finalize;
+
+ factory_class->construct = rtsp_media_factory_wfd_construct;
+ factory_class->create_element = rtsp_media_factory_wfd_create_element;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_media_wfd_debug, "rtspmediafactorywfd", 0,
+ "GstRTSPMediaFactoryWFD");
+}
+
+void gst_rtsp_media_factory_wfd_set (GstRTSPMediaFactoryWFD * factory,
+ guint8 videosrc_type, gchar *audio_device, guint64 audio_latency_time,
+ guint64 audio_buffer_time, gboolean audio_do_timestamp, guint mtu_size)
+{
+ GstRTSPMediaFactoryWFDPrivate *priv =
+ GST_RTSP_MEDIA_FACTORY_WFD_GET_PRIVATE (factory);
+ factory->priv = priv;
+
+ priv->videosrc_type = videosrc_type;
+ priv->audio_device = audio_device;
+ priv->audio_latency_time = audio_latency_time;
+ priv->audio_buffer_time = audio_buffer_time;
+ priv->audio_do_timestamp = audio_do_timestamp;
+ priv->mtu_size = mtu_size;
+}
+
+void gst_rtsp_media_factory_wfd_set_encoders (GstRTSPMediaFactoryWFD * factory,
+ gchar *video_encoder, gchar *audio_encoder_aac, gchar *audio_encoder_ac3)
+{
+ GstRTSPMediaFactoryWFDPrivate *priv =
+ GST_RTSP_MEDIA_FACTORY_WFD_GET_PRIVATE (factory);
+ factory->priv = priv;
+
+ priv->video_encoder = video_encoder;
+ priv->audio_encoder_aac = audio_encoder_aac;
+ priv->audio_encoder_ac3 = audio_encoder_ac3;
+}
+
+void gst_rtsp_media_factory_wfd_set_dump_ts (GstRTSPMediaFactoryWFD * factory,
+ gboolean dump_ts)
+{
+ GstRTSPMediaFactoryWFDPrivate *priv =
+ GST_RTSP_MEDIA_FACTORY_WFD_GET_PRIVATE (factory);
+ factory->priv = priv;
+
+ priv->dump_ts = dump_ts;
+}
+void gst_rtsp_media_factory_wfd_set_negotiated_resolution (GstRTSPMediaFactory *factory,
+ guint32 width, guint32 height)
+{
+ GstRTSPMediaFactoryWFD *factory_wfd = GST_RTSP_MEDIA_FACTORY_WFD (factory);
+ GstRTSPMediaFactoryWFDPrivate *priv = factory_wfd->priv;
+
+ priv->video_width = width;
+ priv->video_height = height;
+}
+void gst_rtsp_media_factory_wfd_set_audio_codec (GstRTSPMediaFactory *factory,
+ guint audio_codec)
+{
+ GstRTSPMediaFactoryWFD *factory_wfd = GST_RTSP_MEDIA_FACTORY_WFD (factory);
+ GstRTSPMediaFactoryWFDPrivate *priv = factory_wfd->priv;
+
+ priv->audio_codec = audio_codec;
+}
+
+static void
+gst_rtsp_media_factory_wfd_init (GstRTSPMediaFactoryWFD * factory)
+{
+ GstRTSPMediaFactoryWFDPrivate *priv =
+ GST_RTSP_MEDIA_FACTORY_WFD_GET_PRIVATE (factory);
+ factory->priv = priv;
+
+ priv->launch = g_strdup (DEFAULT_LAUNCH);
+ priv->shared = DEFAULT_SHARED;
+ priv->protocols = DEFAULT_PROTOCOLS;
+ priv->buffer_size = DEFAULT_BUFFER_SIZE;
+
+ //priv->videosrc_type = GST_WFD_VSRC_XIMAGESRC;
+ //priv->videosrc_type = GST_WFD_VSRC_XVIMAGESRC;
+ //priv->videosrc_type = GST_WFD_VSRC_CAMERASRC;
+ priv->videosrc_type = GST_WFD_VSRC_VIDEOTESTSRC;
+ priv->video_codec = GST_WFD_VIDEO_H264;
+ priv->video_encoder = g_strdup ("omxh264enc");
+ priv->video_bitrate = 200000;
+ priv->video_width = 640;
+ priv->video_height = 480;
+ priv->video_framerate = 30;
+ priv->video_enc_skip_inbuf_value = 5;
+
+ priv->audio_device = g_strdup ("alsa_output.1.analog-stereo.monitor");
+ priv->audio_codec = GST_WFD_AUDIO_AAC;
+ priv->audio_encoder_aac = g_strdup ("avenc_aac");
+ priv->audio_encoder_ac3 = g_strdup ("avenc_ac3");
+ priv->audio_latency_time = 10000;
+ priv->audio_buffer_time = 200000;
+ priv->audio_do_timestamp = FALSE;
+ priv->audio_channels = GST_WFD_CHANNEL_2;
+ priv->audio_freq = GST_WFD_FREQ_48000;
+
+ g_mutex_init (&priv->lock);
+}
+
+static void
+gst_rtsp_media_factory_wfd_finalize (GObject * obj)
+{
+ GstRTSPMediaFactoryWFD *factory = GST_RTSP_MEDIA_FACTORY_WFD (obj);
+ GstRTSPMediaFactoryWFDPrivate *priv = factory->priv;
+
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+ g_free (priv->launch);
+ g_mutex_clear (&priv->lock);
+
+ if (priv->audio_device)
+ g_free (priv->audio_device);
+ if (priv->audio_encoder_aac)
+ g_free (priv->audio_encoder_aac);
+ if (priv->audio_encoder_ac3)
+ g_free (priv->audio_encoder_ac3);
+
+ if (priv->video_encoder)
+ g_free (priv->video_encoder);
+
+ G_OBJECT_CLASS (gst_rtsp_media_factory_wfd_parent_class)->finalize (obj);
+}
+
+GstRTSPMediaFactoryWFD *
+gst_rtsp_media_factory_wfd_new (void)
+{
+ GstRTSPMediaFactoryWFD *result;
+
+ result = g_object_new (GST_TYPE_RTSP_MEDIA_FACTORY_WFD, NULL);
+
+ return result;
+}
+
+static void
+gst_rtsp_media_factory_wfd_get_property (GObject * object,
+ guint propid, GValue * value, GParamSpec * pspec)
+{
+ //GstRTSPMediaFactoryWFD *factory = GST_RTSP_MEDIA_FACTORY_WFD (object);
+
+ switch (propid) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_media_factory_wfd_set_property (GObject * object,
+ guint propid, const GValue * value, GParamSpec * pspec)
+{
+ //GstRTSPMediaFactoryWFD *factory = GST_RTSP_MEDIA_FACTORY_WFD (object);
+
+ switch (propid) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static GstPadProbeReturn
+rtsp_media_wfd_dump_data (GstPad * pad, GstPadProbeInfo *info, gpointer u_data)
+{
+ guint8 *data;
+ gsize size;
+ FILE *f;
+ GstMapInfo mapinfo;
+
+ if (info->type == (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH)) {
+ GstBuffer *buffer = gst_pad_probe_info_get_buffer (info);
+
+ gst_buffer_map (buffer, &mapinfo, GST_MAP_READ);
+ data = mapinfo.data;
+ size = gst_buffer_get_size (buffer);
+
+ f = fopen ("/root/probe.ts", "a");
+ if (f != NULL) {
+ fwrite (data, size, 1, f);
+ fclose (f);
+ }
+ gst_buffer_unmap (buffer, &mapinfo);
+ }
+
+ return GST_PAD_PROBE_OK;
+}
+
+static gboolean
+_rtsp_media_factory_wfd_create_audio_capture_bin (GstRTSPMediaFactoryWFD *
+ factory, GstBin * srcbin)
+{
+ GstElement *audiosrc = NULL;
+ GstElement *acaps = NULL;
+ GstElement *acaps2 = NULL;
+ GstElement *aenc = NULL;
+ GstElement *audio_convert = NULL;
+ GstElement *aqueue = NULL;
+ GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+ guint channels = 0;
+ gboolean is_enc_req = TRUE;
+ guint freq = 0;
+ gchar *acodec = NULL;
+
+ priv = factory->priv;
+
+ /* create audio src element */
+ audiosrc = gst_element_factory_make ("pulsesrc", "audiosrc");
+ if (!audiosrc) {
+ GST_ERROR_OBJECT (factory, "failed to create audiosrc element");
+ goto create_error;
+ }
+
+ GST_INFO_OBJECT (factory, "audio device : %s", priv->audio_device);
+ GST_INFO_OBJECT (factory, "audio latency time : %"G_GUINT64_FORMAT,
+ priv->audio_latency_time);
+ GST_INFO_OBJECT (factory, "audio_buffer_time : %"G_GUINT64_FORMAT,
+ priv->audio_buffer_time);
+ GST_INFO_OBJECT (factory, "audio_do_timestamp : %d",
+ priv->audio_do_timestamp);
+
+ g_object_set (audiosrc, "device", priv->audio_device, NULL);
+ g_object_set (audiosrc, "buffer-time", (gint64) priv->audio_buffer_time,
+ NULL);
+ g_object_set (audiosrc, "latency-time", (gint64) priv->audio_latency_time,
+ NULL);
+ g_object_set (audiosrc, "do-timestamp", (gboolean) priv->audio_do_timestamp,
+ NULL);
+ g_object_set (audiosrc, "provide-clock", (gboolean) FALSE, NULL);
+ g_object_set (audiosrc, "is-live", (gboolean) TRUE, NULL);
+
+ if (priv->audio_codec == GST_WFD_AUDIO_LPCM) {
+ /* To meet miracast certification */
+ gint64 block_size = 1920;
+ g_object_set (audiosrc, "blocksize", (gint64) block_size, NULL);
+
+ audio_convert = gst_element_factory_make ("capssetter", "audio_convert");
+ if (NULL == audio_convert) {
+ GST_ERROR_OBJECT (factory, "failed to create audio convert element");
+ goto create_error;
+ }
+ g_object_set (audio_convert, "caps", gst_caps_new_simple("audio/x-lpcm",
+ "width", G_TYPE_INT, 16,
+ "rate", G_TYPE_INT, 48000,
+ "channels", G_TYPE_INT, 2,
+ "dynamic_range", G_TYPE_INT, 0,
+ "emphasis", G_TYPE_BOOLEAN, FALSE,
+ "mute", G_TYPE_BOOLEAN, FALSE, NULL), NULL);
+ g_object_set (audio_convert, "join", (gboolean)FALSE, NULL);
+ g_object_set (audio_convert, "replace", (gboolean)TRUE, NULL);
+
+ acaps2 = gst_element_factory_make ("capsfilter", "audiocaps2");
+ if (NULL == acaps2) {
+ GST_ERROR_OBJECT (factory, "failed to create audio capsilfter element");
+ goto create_error;
+ }
+ /* In case of LPCM, uses big endian */
+ g_object_set (G_OBJECT (acaps2), "caps",
+ gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S16BE",
+ /* In case of LPCM, uses big endian */
+ "rate", G_TYPE_INT, 48000,
+ "channels", G_TYPE_INT, 2, NULL), NULL);
+ }
+
+ /* create audio caps element */
+ acaps = gst_element_factory_make ("capsfilter", "audiocaps");
+ if (NULL == acaps) {
+ GST_ERROR_OBJECT (factory, "failed to create audio capsilfter element");
+ goto create_error;
+ }
+
+ if (priv->audio_channels == GST_WFD_CHANNEL_2)
+ channels = 2;
+ else if (priv->audio_channels == GST_WFD_CHANNEL_4)
+ channels = 4;
+ else if (priv->audio_channels == GST_WFD_CHANNEL_6)
+ channels = 6;
+ else if (priv->audio_channels == GST_WFD_CHANNEL_8)
+ channels = 8;
+ else
+ channels = 2;
+
+ if (priv->audio_freq == GST_WFD_FREQ_44100)
+ freq = 44100;
+ else if (priv->audio_freq == GST_WFD_FREQ_48000)
+ freq = 48000;
+ else
+ freq = 44100;
+
+ if (priv->audio_codec == GST_WFD_AUDIO_LPCM) {
+ g_object_set (G_OBJECT (acaps), "caps",
+ gst_caps_new_simple ("audio/x-lpcm", "width", G_TYPE_INT, 16,
+ "rate", G_TYPE_INT, 48000,
+ "channels", G_TYPE_INT, 2,
+ "dynamic_range", G_TYPE_INT, 0,
+ "emphasis", G_TYPE_BOOLEAN, FALSE,
+ "mute", G_TYPE_BOOLEAN, FALSE, NULL), NULL);
+ } else if ((priv->audio_codec == GST_WFD_AUDIO_AAC)
+ || (priv->audio_codec == GST_WFD_AUDIO_AC3)) {
+ g_object_set (G_OBJECT (acaps), "caps", gst_caps_new_simple ("audio/x-raw",
+ "endianness", G_TYPE_INT, 1234, "signed", G_TYPE_BOOLEAN, TRUE,
+ "depth", G_TYPE_INT, 16, "rate", G_TYPE_INT, freq, "channels",
+ G_TYPE_INT, channels, NULL), NULL);
+ }
+
+ if (priv->audio_codec == GST_WFD_AUDIO_AAC) {
+ acodec = g_strdup (priv->audio_encoder_aac);
+ is_enc_req = TRUE;
+ } else if (priv->audio_codec == GST_WFD_AUDIO_AC3) {
+ acodec = g_strdup (priv->audio_encoder_ac3);
+ is_enc_req = TRUE;
+ } else if (priv->audio_codec == GST_WFD_AUDIO_LPCM) {
+ GST_DEBUG_OBJECT (factory, "No codec required, raw data will be sent");
+ is_enc_req = FALSE;
+ } else {
+ GST_ERROR_OBJECT (factory, "Yet to support other than H264 format");
+ goto create_error;
+ }
+
+ if (is_enc_req) {
+ aenc = gst_element_factory_make (acodec, "audioenc");
+ if (NULL == aenc) {
+ GST_ERROR_OBJECT (factory, "failed to create audio encoder element");
+ goto create_error;
+ }
+
+ g_object_set (aenc, "compliance", -2, NULL);
+ g_object_set (aenc, "tolerance", 400000000, NULL);
+ g_object_set (aenc, "bitrate", (guint) 128000, NULL);
+ g_object_set (aenc, "rate-control", 2, NULL);
+
+ aqueue = gst_element_factory_make ("queue", "audio-queue");
+ if (!aqueue) {
+ GST_ERROR_OBJECT (factory, "failed to create audio queue element");
+ goto create_error;
+ }
+
+ gst_bin_add_many (srcbin, audiosrc, acaps, aenc, aqueue, NULL);
+
+ if (!gst_element_link_many (audiosrc, acaps, aenc, aqueue, NULL)) {
+ GST_ERROR_OBJECT (factory, "Failed to link audio src elements...");
+ goto create_error;
+ }
+ } else {
+ aqueue = gst_element_factory_make ("queue", "audio-queue");
+ if (!aqueue) {
+ GST_ERROR_OBJECT (factory, "failed to create audio queue element");
+ goto create_error;
+ }
+
+ gst_bin_add_many (srcbin, audiosrc, acaps2, audio_convert, acaps, aqueue, NULL);
+
+ if (!gst_element_link_many (audiosrc, acaps2, audio_convert, acaps, aqueue, NULL)) {
+ GST_ERROR_OBJECT (factory, "Failed to link audio src elements...");
+ goto create_error;
+ }
+ }
+
+ priv->audio_queue = aqueue;
+ if (acodec) g_free (acodec);
+
+ return TRUE;
+
+create_error:
+ if (acodec) g_free (acodec);
+ return FALSE;
+}
+
+static gboolean
+_rtsp_media_factory_wfd_create_videotest_bin (GstRTSPMediaFactoryWFD * factory,
+ GstBin * srcbin)
+{
+ GstElement *videosrc = NULL;
+ GstElement *vcaps = NULL;
+ GstElement *videoconvert = NULL;
+ GstElement *venc_caps = NULL;
+ gchar *vcodec = NULL;
+ GstElement *venc = NULL;
+ GstElement *vparse = NULL;
+ GstElement *vqueue = NULL;
+ GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+ priv = factory->priv;
+
+ GST_INFO_OBJECT (factory, "picked videotestsrc as video source");
+
+ videosrc = gst_element_factory_make ("videotestsrc", "videosrc");
+ if (NULL == videosrc) {
+ GST_ERROR_OBJECT (factory, "failed to create ximagesrc element");
+ goto create_error;
+ }
+
+ /* create video caps element */
+ vcaps = gst_element_factory_make ("capsfilter", "videocaps");
+ if (NULL == vcaps) {
+ GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+ goto create_error;
+ }
+
+ g_object_set (G_OBJECT (vcaps), "caps",
+ gst_caps_new_simple ("video/x-raw",
+ "format", G_TYPE_STRING, "I420",
+ "width", G_TYPE_INT, priv->video_width,
+ "height", G_TYPE_INT, priv->video_height,
+ "framerate", GST_TYPE_FRACTION, priv->video_framerate, 1, NULL),
+ NULL);
+
+ /* create video convert element */
+ videoconvert = gst_element_factory_make ("videoconvert", "videoconvert");
+ if (NULL == videoconvert) {
+ GST_ERROR_OBJECT (factory, "failed to create video videoconvert element");
+ goto create_error;
+ }
+
+ venc_caps = gst_element_factory_make ("capsfilter", "venc_caps");
+ if (NULL == venc_caps) {
+ GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+ goto create_error;
+ }
+
+ g_object_set (G_OBJECT (venc_caps), "caps",
+ gst_caps_new_simple ("video/x-raw",
+ "format", G_TYPE_STRING, "SN12",
+ "width", G_TYPE_INT, priv->video_width,
+ "height", G_TYPE_INT, priv->video_height,
+ "framerate", GST_TYPE_FRACTION, priv->video_framerate, 1, NULL),
+ NULL);
+
+ if (priv->video_codec == GST_WFD_VIDEO_H264)
+ vcodec = g_strdup (priv->video_encoder);
+ else {
+ GST_ERROR_OBJECT (factory, "Yet to support other than H264 format");
+ goto create_error;
+ }
+
+ venc = gst_element_factory_make (vcodec, "videoenc");
+ if (vcodec) g_free (vcodec);
+
+ if (!venc) {
+ GST_ERROR_OBJECT (factory, "failed to create video encoder element");
+ goto create_error;
+ }
+
+ g_object_set (venc, "aud", 0, NULL);
+ g_object_set (venc, "byte-stream", 1, NULL);
+ g_object_set (venc, "bitrate", 512, NULL);
+
+ vparse = gst_element_factory_make ("h264parse", "videoparse");
+ if (NULL == vparse) {
+ GST_ERROR_OBJECT (factory, "failed to create h264 parse element");
+ goto create_error;
+ }
+ g_object_set (vparse, "config-interval", 1, NULL);
+
+ vqueue = gst_element_factory_make ("queue", "video-queue");
+ if (!vqueue) {
+ GST_ERROR_OBJECT (factory, "failed to create video queue element");
+ goto create_error;
+ }
+
+ gst_bin_add_many (srcbin, videosrc, vcaps, videoconvert, venc_caps, venc, vparse, vqueue, NULL);
+ if (!gst_element_link_many (videosrc, vcaps, videoconvert, venc_caps, venc, vparse, vqueue, NULL)) {
+ GST_ERROR_OBJECT (factory, "Failed to link video src elements...");
+ goto create_error;
+ }
+
+ priv->video_queue = vqueue;
+
+ return TRUE;
+
+create_error:
+ return FALSE;
+}
+
+static gboolean
+_rtsp_media_factory_wfd_create_camera_capture_bin (GstRTSPMediaFactoryWFD *
+ factory, GstBin * srcbin)
+{
+ GstElement *videosrc = NULL;
+ GstElement *vcaps = NULL;
+ GstElement *venc = NULL;
+ GstElement *vparse = NULL;
+ GstElement *vqueue = NULL;
+ gchar *vcodec = NULL;
+ GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+ priv = factory->priv;
+
+ videosrc = gst_element_factory_make ("camerasrc", "videosrc");
+ if (NULL == videosrc) {
+ GST_ERROR_OBJECT (factory, "failed to create camerasrc element");
+ goto create_error;
+ }
+
+ /* create video caps element */
+ vcaps = gst_element_factory_make ("capsfilter", "videocaps");
+ if (NULL == vcaps) {
+ GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+ goto create_error;
+ }
+
+ GST_INFO_OBJECT (factory, "picked camerasrc as video source");
+ g_object_set (G_OBJECT (vcaps), "caps",
+ gst_caps_new_simple ("video/x-raw",
+ "width", G_TYPE_INT, priv->video_width,
+ "height", G_TYPE_INT, priv->video_height,
+ "format", G_TYPE_STRING, "SN12",
+ "framerate", GST_TYPE_FRACTION, priv->video_framerate, 1, NULL),
+ NULL);
+
+ if (priv->video_codec == GST_WFD_VIDEO_H264)
+ vcodec = g_strdup (priv->video_encoder);
+ else {
+ GST_ERROR_OBJECT (factory, "Yet to support other than H264 format");
+ goto create_error;
+ }
+
+ venc = gst_element_factory_make (vcodec, "videoenc");
+ if (!venc) {
+ GST_ERROR_OBJECT (factory, "failed to create video encoder element");
+ goto create_error;
+ }
+ if (vcodec) g_free (vcodec);
+
+ g_object_set (venc, "bitrate", priv->video_bitrate, NULL);
+ g_object_set (venc, "byte-stream", 1, NULL);
+ g_object_set (venc, "append-dci", 1, NULL);
+
+ vparse = gst_element_factory_make ("h264parse", "videoparse");
+ if (NULL == vparse) {
+ GST_ERROR_OBJECT (factory, "failed to create h264 parse element");
+ goto create_error;
+ }
+ g_object_set (vparse, "config-interval", 1, NULL);
+
+ vqueue = gst_element_factory_make ("queue", "video-queue");
+ if (!vqueue) {
+ GST_ERROR_OBJECT (factory, "failed to create video queue element");
+ goto create_error;
+ }
+
+ gst_bin_add_many (srcbin, videosrc, vcaps, venc, vparse, vqueue, NULL);
+
+ if (!gst_element_link_many (videosrc, vcaps, venc, vparse, vqueue, NULL)) {
+ GST_ERROR_OBJECT (factory, "Failed to link video src elements...");
+ goto create_error;
+ }
+
+ priv->video_queue = vqueue;
+
+ return TRUE;
+
+create_error:
+ return FALSE;
+}
+
+static gboolean
+_rtsp_media_factory_wfd_create_xcapture_bin (GstRTSPMediaFactoryWFD * factory,
+ GstBin * srcbin)
+{
+ GstElement *videosrc = NULL;
+ GstElement *vcaps = NULL;
+ GstElement *venc_caps = NULL;
+ GstElement *videoconvert = NULL, *videoscale = NULL;
+ gchar *vcodec = NULL;
+ GstElement *venc = NULL;
+ GstElement *vparse = NULL;
+ GstElement *vqueue = NULL;
+ GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+ priv = factory->priv;
+
+ GST_INFO_OBJECT (factory, "picked ximagesrc as video source");
+
+ videosrc = gst_element_factory_make ("ximagesrc", "videosrc");
+ if (NULL == videosrc) {
+ GST_ERROR_OBJECT (factory, "failed to create ximagesrc element");
+ goto create_error;
+ }
+
+ videoscale = gst_element_factory_make ("videoscale", "videoscale");
+ if (NULL == videoscale) {
+ GST_ERROR_OBJECT (factory, "failed to create videoscale element");
+ goto create_error;
+ }
+
+ videoconvert = gst_element_factory_make ("videoconvert", "videoconvert");
+ if (NULL == videoconvert) {
+ GST_ERROR_OBJECT (factory, "failed to create videoconvert element");
+ goto create_error;
+ }
+
+ /* create video caps element */
+ vcaps = gst_element_factory_make ("capsfilter", "videocaps");
+ if (NULL == vcaps) {
+ GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+ goto create_error;
+ }
+
+ g_object_set (G_OBJECT (vcaps), "caps",
+ gst_caps_new_simple ("video/x-raw",
+ "width", G_TYPE_INT, priv->video_width,
+ "height", G_TYPE_INT, priv->video_height,
+ "framerate", GST_TYPE_FRACTION, priv->video_framerate, 1, NULL),
+ NULL);
+
+ if (priv->video_codec == GST_WFD_VIDEO_H264)
+ vcodec = g_strdup (priv->video_encoder);
+ else {
+ GST_ERROR_OBJECT (factory, "Yet to support other than H264 format");
+ goto create_error;
+ }
+
+ venc = gst_element_factory_make (vcodec, "videoenc");
+ if (vcodec) g_free (vcodec);
+
+ if (!venc) {
+ GST_ERROR_OBJECT (factory, "failed to create video encoder element");
+ goto create_error;
+ }
+
+ g_object_set (venc, "aud", 0, NULL);
+ g_object_set (venc, "byte-stream", 1, NULL);
+ g_object_set (venc, "bitrate", 512, NULL);
+
+ venc_caps = gst_element_factory_make ("capsfilter", "venc_caps");
+ if (NULL == venc_caps) {
+ GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+ goto create_error;
+ }
+
+ g_object_set (G_OBJECT (venc_caps), "caps",
+ gst_caps_new_simple ("video/x-h264",
+ "profile", G_TYPE_STRING, "baseline", NULL), NULL);
+
+ vparse = gst_element_factory_make ("h264parse", "videoparse");
+ if (NULL == vparse) {
+ GST_ERROR_OBJECT (factory, "failed to create h264 parse element");
+ goto create_error;
+ }
+ g_object_set (vparse, "config-interval", 1, NULL);
+
+ vqueue = gst_element_factory_make ("queue", "video-queue");
+ if (!vqueue) {
+ GST_ERROR_OBJECT (factory, "failed to create video queue element");
+ goto create_error;
+ }
+
+ gst_bin_add_many (srcbin, videosrc, videoscale, videoconvert, vcaps, venc,
+ venc_caps, vparse, vqueue, NULL);
+ if (!gst_element_link_many (videosrc, videoscale, videoconvert, vcaps, venc,
+ venc_caps, vparse, vqueue, NULL)) {
+ GST_ERROR_OBJECT (factory, "Failed to link video src elements...");
+ goto create_error;
+ }
+
+ priv->video_queue = vqueue;
+
+ return TRUE;
+
+create_error:
+ return FALSE;
+}
+
+static gboolean
+_rtsp_media_factory_wfd_create_xvcapture_bin (GstRTSPMediaFactoryWFD * factory,
+ GstBin * srcbin)
+{
+ GstElement *videosrc = NULL;
+ GstElement *vcaps = NULL;
+ gchar *vcodec = NULL;
+ GstElement *venc = NULL;
+ GstElement *vparse = NULL;
+ GstElement *vqueue = NULL;
+ GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+ priv = factory->priv;
+
+ GST_INFO_OBJECT (factory, "picked xvimagesrc as video source");
+
+ videosrc = gst_element_factory_make ("xvimagesrc", "videosrc");
+ if (NULL == videosrc) {
+ GST_ERROR_OBJECT (factory, "failed to create xvimagesrc element");
+ goto create_error;
+ }
+
+ /* create video caps element */
+ vcaps = gst_element_factory_make ("capsfilter", "videocaps");
+ if (NULL == vcaps) {
+ GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+ goto create_error;
+ }
+
+ g_object_set (G_OBJECT (vcaps), "caps",
+ gst_caps_new_simple ("video/x-raw",
+ "width", G_TYPE_INT, priv->video_width,
+ "height", G_TYPE_INT, priv->video_height,
+ "format", G_TYPE_STRING, "SN12",
+ "framerate", GST_TYPE_FRACTION, priv->video_framerate, 1, NULL),
+ NULL);
+
+ if (priv->video_codec == GST_WFD_VIDEO_H264) {
+ vcodec = g_strdup (priv->video_encoder);
+ } else {
+ GST_ERROR_OBJECT (factory, "Yet to support other than H264 format");
+ goto create_error;
+ }
+
+ venc = gst_element_factory_make (vcodec, "videoenc");
+ if (!venc) {
+ GST_ERROR_OBJECT (factory, "failed to create video encoder element");
+ goto create_error;
+ }
+ g_object_set (venc, "bitrate", priv->video_bitrate, NULL);
+ g_object_set (venc, "byte-stream", 1, NULL);
+ g_object_set (venc, "append-dci", 1, NULL);
+ g_object_set (venc, "idr-period", 120, NULL);
+ g_object_set (venc, "skip-inbuf", priv->video_enc_skip_inbuf_value, NULL);
+
+ vparse = gst_element_factory_make ("h264parse", "videoparse");
+ if (NULL == vparse) {
+ GST_ERROR_OBJECT (factory, "failed to create h264 parse element");
+ goto create_error;
+ }
+ g_object_set (vparse, "config-interval", 1, NULL);
+
+ vqueue = gst_element_factory_make ("queue", "video-queue");
+ if (!vqueue) {
+ GST_ERROR_OBJECT (factory, "failed to create video queue element");
+ goto create_error;
+ }
+
+ gst_bin_add_many (srcbin, videosrc, vcaps, venc, vparse, vqueue, NULL);
+ if (!gst_element_link_many (videosrc, vcaps, venc, vparse, vqueue, NULL)) {
+ GST_ERROR_OBJECT (factory, "Failed to link video src elements...");
+ goto create_error;
+ }
+
+ priv->video_queue = vqueue;
+ if (vcodec) g_free (vcodec);
+
+ return TRUE;
+
+create_error:
+ if (vcodec) g_free (vcodec);
+ return FALSE;
+}
+
+static GstElement *
+_rtsp_media_factory_wfd_create_srcbin (GstRTSPMediaFactoryWFD * factory)
+{
+ GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+ GstBin *srcbin = NULL;
+ GstElement *mux = NULL;
+ GstElement *mux_queue = NULL;
+ GstElement *payload = NULL;
+ GstPad *srcpad = NULL;
+ GstPad *mux_vsinkpad = NULL;
+ GstPad *mux_asinkpad = NULL;
+
+ priv = factory->priv;
+
+ /* create source bin */
+ srcbin = GST_BIN (gst_bin_new ("srcbin"));
+ if (!srcbin) {
+ GST_ERROR_OBJECT (factory, "failed to create source bin...");
+ goto create_error;
+ }
+
+ /* create video src element */
+ switch (priv->videosrc_type) {
+ case GST_WFD_VSRC_XIMAGESRC:
+ if (!_rtsp_media_factory_wfd_create_xcapture_bin (factory, srcbin)) {
+ GST_ERROR_OBJECT (factory, "failed to create xcapture bin...");
+ goto create_error;
+ }
+ break;
+ case GST_WFD_VSRC_XVIMAGESRC:
+ if (!_rtsp_media_factory_wfd_create_xvcapture_bin (factory, srcbin)) {
+ GST_ERROR_OBJECT (factory, "failed to create xvcapture bin...");
+ goto create_error;
+ }
+ break;
+ case GST_WFD_VSRC_CAMERASRC:
+ if (!_rtsp_media_factory_wfd_create_camera_capture_bin (factory, srcbin)) {
+ GST_ERROR_OBJECT (factory, "failed to create camera capture bin...");
+ goto create_error;
+ }
+ break;
+ case GST_WFD_VSRC_VIDEOTESTSRC:
+ if (!_rtsp_media_factory_wfd_create_videotest_bin (factory, srcbin)) {
+ GST_ERROR_OBJECT (factory, "failed to create videotestsrc bin...");
+ goto create_error;
+ }
+ break;
+ default:
+ GST_ERROR_OBJECT (factory, "unknow mode selected...");
+ goto create_error;
+ }
+
+ mux = gst_element_factory_make ("mpegtsmux", "tsmux");
+ if (!mux) {
+ GST_ERROR_OBJECT (factory, "failed to create muxer element");
+ goto create_error;
+ }
+
+ g_object_set (mux, "wfd-mode", TRUE, NULL);
+
+ mux_queue = gst_element_factory_make ("queue", "muxer-queue");
+ if (!mux_queue) {
+ GST_ERROR_OBJECT (factory, "failed to create muxer-queue element");
+ goto create_error;
+ }
+
+ g_object_set (mux_queue, "max-size-buffers", 20000, NULL);
+
+ payload = gst_element_factory_make ("rtpmp2tpay", "pay0");
+ if (!payload) {
+ GST_ERROR_OBJECT (factory, "failed to create payload element");
+ goto create_error;
+ }
+
+ g_object_set (payload, "pt", 33, NULL);
+ g_object_set (payload, "mtu", priv->mtu_size, NULL);
+ g_object_set (payload, "rtp-flush", (gboolean) TRUE, NULL);
+
+ gst_bin_add_many (srcbin, mux, mux_queue, payload, NULL);
+
+ if (!gst_element_link_many (mux, mux_queue, payload, NULL)) {
+ GST_ERROR_OBJECT (factory, "Failed to link muxer & payload...");
+ goto create_error;
+ }
+
+ /* request video sink pad from muxer, which has elementary pid 0x1011 */
+ mux_vsinkpad = gst_element_get_request_pad (mux, "sink_4113");
+ if (!mux_vsinkpad) {
+ GST_ERROR_OBJECT (factory, "Failed to get sink pad from muxer...");
+ goto create_error;
+ }
+
+ /* request srcpad from video queue */
+ srcpad = gst_element_get_static_pad (priv->video_queue, "src");
+ if (!srcpad) {
+ GST_ERROR_OBJECT (factory, "Failed to get srcpad from video queue...");
+ goto create_error;
+ }
+
+ if (gst_pad_link (srcpad, mux_vsinkpad) != GST_PAD_LINK_OK) {
+ GST_ERROR_OBJECT (factory,
+ "Failed to link video queue src pad & muxer video sink pad...");
+ goto create_error;
+ }
+
+ gst_object_unref (mux_vsinkpad);
+ gst_object_unref (srcpad);
+ srcpad = NULL;
+
+ /* create audio source elements & add to pipeline */
+ if (!_rtsp_media_factory_wfd_create_audio_capture_bin (factory, srcbin))
+ goto create_error;
+
+ /* request audio sink pad from muxer, which has elementary pid 0x1100 */
+ mux_asinkpad = gst_element_get_request_pad (mux, "sink_4352");
+ if (!mux_asinkpad) {
+ GST_ERROR_OBJECT (factory, "Failed to get sinkpad from muxer...");
+ goto create_error;
+ }
+
+ /* request srcpad from audio queue */
+ srcpad = gst_element_get_static_pad (priv->audio_queue, "src");
+ if (!srcpad) {
+ GST_ERROR_OBJECT (factory, "Failed to get srcpad from audio queue...");
+ goto create_error;
+ }
+
+ /* link audio queue's srcpad & muxer sink pad */
+ if (gst_pad_link (srcpad, mux_asinkpad) != GST_PAD_LINK_OK) {
+ GST_ERROR_OBJECT (factory,
+ "Failed to link audio queue src pad & muxer audio sink pad...");
+ goto create_error;
+ }
+ gst_object_unref (mux_asinkpad);
+ gst_object_unref (srcpad);
+
+ if (priv->dump_ts)
+ {
+ GstPad *pad_probe = NULL;
+ pad_probe = gst_element_get_static_pad (mux, "src");
+
+ if (NULL == pad_probe) {
+ GST_INFO_OBJECT (factory, "pad for probe not created");
+ } else {
+ GST_INFO_OBJECT (factory, "pad for probe SUCCESSFUL");
+ }
+ gst_pad_add_probe (pad_probe, GST_PAD_PROBE_TYPE_BUFFER,
+ rtsp_media_wfd_dump_data, factory, NULL);
+ }
+
+ GST_DEBUG_OBJECT (factory, "successfully created source bin...");
+
+ return GST_ELEMENT_CAST (srcbin);
+
+create_error:
+ GST_ERROR_OBJECT (factory, "Failed to create pipeline");
+ return NULL;
+}
+
+static GstElement *
+rtsp_media_factory_wfd_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMediaFactoryWFD *_factory = GST_RTSP_MEDIA_FACTORY_WFD_CAST (factory);
+ GstElement *element = NULL;
+
+ GST_RTSP_MEDIA_FACTORY_WFD_LOCK (factory);
+
+ element = _rtsp_media_factory_wfd_create_srcbin (_factory);
+
+ GST_RTSP_MEDIA_FACTORY_WFD_UNLOCK (factory);
+
+ return element;
+}
+
+static GstRTSPMedia *
+rtsp_media_factory_wfd_construct (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMedia *media;
+ GstElement *element, *pipeline;
+ GstRTSPMediaFactoryClass *klass;
+
+ klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+ if (!klass->create_pipeline)
+ goto no_create;
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ if (element == NULL)
+ goto no_element;
+
+ /* create a new empty media */
+ media = gst_rtsp_media_new (element);
+
+ gst_rtsp_media_collect_streams (media);
+
+ pipeline = klass->create_pipeline (factory, media);
+ if (pipeline == NULL)
+ goto no_pipeline;
+
+ return media;
+
+ /* ERRORS */
+no_create:
+ {
+ g_critical ("no create_pipeline function");
+ return NULL;
+ }
+no_element:
+ {
+ g_critical ("could not create element");
+ return NULL;
+ }
+no_pipeline:
+ {
+ g_critical ("can't create pipeline");
+ g_object_unref (media);
+ return NULL;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include "rtsp-media-factory.h"
+
+#ifndef __GST_RTSP_MEDIA_FACTORY_WFD_H__
+#define __GST_RTSP_MEDIA_FACTORY_WFD_H__
+
+G_BEGIN_DECLS
+/* types for the media factory */
+#define GST_TYPE_RTSP_MEDIA_FACTORY_WFD (gst_rtsp_media_factory_wfd_get_type ())
+#define GST_IS_RTSP_MEDIA_FACTORY_WFD(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_WFD))
+#define GST_IS_RTSP_MEDIA_FACTORY_WFD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA_FACTORY_WFD))
+#define GST_RTSP_MEDIA_FACTORY_WFD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_WFD, GstRTSPMediaFactoryWFDClass))
+#define GST_RTSP_MEDIA_FACTORY_WFD(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_WFD, GstRTSPMediaFactoryWFD))
+#define GST_RTSP_MEDIA_FACTORY_WFD_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA_FACTORY_WFD, GstRTSPMediaFactoryWFDClass))
+#define GST_RTSP_MEDIA_FACTORY_WFD_CAST(obj) ((GstRTSPMediaFactoryWFD*)(obj))
+#define GST_RTSP_MEDIA_FACTORY_WFD_CLASS_CAST(klass) ((GstRTSPMediaFactoryWFDClass*)(klass))
+ enum
+{
+ GST_WFD_VSRC_XIMAGESRC,
+ GST_WFD_VSRC_XVIMAGESRC,
+ GST_WFD_VSRC_CAMERASRC,
+ GST_WFD_VSRC_VIDEOTESTSRC
+};
+
+typedef struct _GstRTSPMediaFactoryWFD GstRTSPMediaFactoryWFD;
+typedef struct _GstRTSPMediaFactoryWFDClass GstRTSPMediaFactoryWFDClass;
+typedef struct _GstRTSPMediaFactoryWFDPrivate GstRTSPMediaFactoryWFDPrivate;
+
+/**
+ * GstRTSPMediaFactoryWFD:
+ *
+ * The definition and logic for constructing the pipeline for a media. The media
+ * can contain multiple streams like audio and video.
+ */
+struct _GstRTSPMediaFactoryWFD
+{
+ GstRTSPMediaFactory parent;
+
+ /*< private > */
+ GstRTSPMediaFactoryWFDPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaFactoryWFDClass:
+ * @gen_key: convert @url to a key for caching shared #GstRTSPMedia objects.
+ * The default implementation of this function will use the complete URL
+ * including the query parameters to return a key.
+ * @create_element: Construct and return a #GstElement that is a #GstBin containing
+ * the elements to use for streaming the media. The bin should contain
+ * payloaders pay\%d for each stream. The default implementation of this
+ * function returns the bin created from the launch parameter.
+ * @construct: the vmethod that will be called when the factory has to create the
+ * #GstRTSPMedia for @url. The default implementation of this
+ * function calls create_element to retrieve an element and then looks for
+ * pay\%d to create the streams.
+ * @create_pipeline: create a new pipeline or re-use an existing one and
+ * add the #GstRTSPMedia's element created by @construct to the pipeline.
+ * @configure: configure the media created with @construct. The default
+ * implementation will configure the 'shared' property of the media.
+ * @media_constructed: signal emited when a media was constructed
+ * @media_configure: signal emited when a media should be configured
+ *
+ * The #GstRTSPMediaFactoryWFD class structure.
+ */
+struct _GstRTSPMediaFactoryWFDClass
+{
+ GstRTSPMediaFactoryClass parent_class;
+
+ gchar *(*gen_key) (GstRTSPMediaFactoryWFD * factory, const GstRTSPUrl * url);
+
+ GstElement *(*create_element) (GstRTSPMediaFactoryWFD * factory,
+ const GstRTSPUrl * url);
+ GstRTSPMedia *(*construct) (GstRTSPMediaFactoryWFD * factory,
+ const GstRTSPUrl * url);
+ GstElement *(*create_pipeline) (GstRTSPMediaFactoryWFD * factory,
+ GstRTSPMedia * media);
+ void (*configure) (GstRTSPMediaFactoryWFD * factory, GstRTSPMedia * media);
+
+ /* signals */
+ void (*media_constructed) (GstRTSPMediaFactoryWFD * factory,
+ GstRTSPMedia * media);
+ void (*media_configure) (GstRTSPMediaFactoryWFD * factory,
+ GstRTSPMedia * media);
+
+ /*< private > */
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType gst_rtsp_media_factory_wfd_get_type (void);
+
+/* creating the factory */
+GstRTSPMediaFactoryWFD *gst_rtsp_media_factory_wfd_new (void);
+GstElement *gst_rtsp_media_factory_wfd_create_element (GstRTSPMediaFactoryWFD *
+ factory, const GstRTSPUrl * url);
+
+void gst_rtsp_media_factory_wfd_set (GstRTSPMediaFactoryWFD * factory,
+ guint8 videosrc_type, gchar *audio_device, guint64 audio_latency_time,
+ guint64 audio_buffer_time, gboolean audio_do_timestamp, guint mtu_size);
+void gst_rtsp_media_factory_wfd_set_encoders (GstRTSPMediaFactoryWFD * factory,
+ gchar *video_encoder, gchar *audio_encoder_aac, gchar *audio_encoder_ac3);
+void gst_rtsp_media_factory_wfd_set_dump_ts (GstRTSPMediaFactoryWFD * factory,
+ gboolean dump_ts);
+void gst_rtsp_media_factory_wfd_set_negotiated_resolution (GstRTSPMediaFactory *factory,
+ guint32 width, guint32 height);
+void gst_rtsp_media_factory_wfd_set_audio_codec (GstRTSPMediaFactory *factory,
+ guint audio_codec);
+
+G_END_DECLS
+#endif /* __GST_RTSP_MEDIA_FACTORY_WFD_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media-factory
+ * @short_description: A factory for media pipelines
+ * @see_also: #GstRTSPMountPoints, #GstRTSPMedia
+ *
+ * The #GstRTSPMediaFactory is responsible for creating or recycling
+ * #GstRTSPMedia objects based on the passed URL.
+ *
+ * The default implementation of the object can create #GstRTSPMedia objects
+ * containing a pipeline created from a launch description set with
+ * gst_rtsp_media_factory_set_launch().
+ *
+ * Media from a factory can be shared by setting the shared flag with
+ * gst_rtsp_media_factory_set_shared(). When a factory is shared,
+ * gst_rtsp_media_factory_construct() will return the same #GstRTSPMedia when
+ * the url matches.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include "rtsp-media-factory.h"
+
+#define GST_RTSP_MEDIA_FACTORY_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactoryPrivate))
+
+#define GST_RTSP_MEDIA_FACTORY_GET_LOCK(f) (&(GST_RTSP_MEDIA_FACTORY_CAST(f)->priv->lock))
+#define GST_RTSP_MEDIA_FACTORY_LOCK(f) (g_mutex_lock(GST_RTSP_MEDIA_FACTORY_GET_LOCK(f)))
+#define GST_RTSP_MEDIA_FACTORY_UNLOCK(f) (g_mutex_unlock(GST_RTSP_MEDIA_FACTORY_GET_LOCK(f)))
+
+struct _GstRTSPMediaFactoryPrivate
+{
+ GMutex lock; /* protects everything but medias */
+ GstRTSPPermissions *permissions;
+ gchar *launch;
+ gboolean shared;
+ GstRTSPSuspendMode suspend_mode;
+ gboolean eos_shutdown;
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans protocols;
+ guint buffer_size;
+ GstRTSPAddressPool *pool;
+
+ GMutex medias_lock;
+ GHashTable *medias; /* protected by medias_lock */
+};
+
+#define DEFAULT_LAUNCH NULL
+#define DEFAULT_SHARED FALSE
+#define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
+#define DEFAULT_EOS_SHUTDOWN FALSE
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+ GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_BUFFER_SIZE 0x80000
+
+enum
+{
+ PROP_0,
+ PROP_LAUNCH,
+ PROP_SHARED,
+ PROP_SUSPEND_MODE,
+ PROP_EOS_SHUTDOWN,
+ PROP_PROFILES,
+ PROP_PROTOCOLS,
+ PROP_BUFFER_SIZE,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_MEDIA_CONSTRUCTED,
+ SIGNAL_MEDIA_CONFIGURE,
+ SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
+#define GST_CAT_DEFAULT rtsp_media_debug
+
+static guint gst_rtsp_media_factory_signals[SIGNAL_LAST] = { 0 };
+
+static void gst_rtsp_media_factory_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_finalize (GObject * obj);
+
+static gchar *default_gen_key (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url);
+static GstElement *default_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url);
+static GstRTSPMedia *default_construct (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url);
+static void default_configure (GstRTSPMediaFactory * factory,
+ GstRTSPMedia * media);
+static GstElement *default_create_pipeline (GstRTSPMediaFactory * factory,
+ GstRTSPMedia * media);
+
+G_DEFINE_TYPE (GstRTSPMediaFactory, gst_rtsp_media_factory, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPMediaFactoryPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_media_factory_get_property;
+ gobject_class->set_property = gst_rtsp_media_factory_set_property;
+ gobject_class->finalize = gst_rtsp_media_factory_finalize;
+
+ /**
+ * GstRTSPMediaFactory::launch:
+ *
+ * The gst_parse_launch() line to use for constructing the pipeline in the
+ * default prepare vmethod.
+ *
+ * The pipeline description should return a GstBin as the toplevel element
+ * which can be accomplished by enclosing the dscription with brackets '('
+ * ')'.
+ *
+ * The description should return a pipeline with payloaders named pay0, pay1,
+ * etc.. Each of the payloaders will result in a stream.
+ *
+ * Support for dynamic payloaders can be accomplished by adding payloaders
+ * named dynpay0, dynpay1, etc..
+ */
+ g_object_class_install_property (gobject_class, PROP_LAUNCH,
+ g_param_spec_string ("launch", "Launch",
+ "A launch description of the pipeline", DEFAULT_LAUNCH,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_SHARED,
+ g_param_spec_boolean ("shared", "Shared",
+ "If media from this factory is shared", DEFAULT_SHARED,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
+ g_param_spec_enum ("suspend-mode", "Suspend Mode",
+ "Control how media will be suspended", GST_TYPE_RTSP_SUSPEND_MODE,
+ DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
+ g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
+ "Send EOS down the pipeline before shutting down",
+ DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
+ g_param_spec_uint ("buffer-size", "Buffer Size",
+ "The kernel UDP buffer size to use", 0, G_MAXUINT,
+ DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONSTRUCTED] =
+ g_signal_new ("media-constructed", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaFactoryClass,
+ media_constructed), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_MEDIA);
+
+ gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONFIGURE] =
+ g_signal_new ("media-configure", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaFactoryClass,
+ media_configure), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_MEDIA);
+
+ klass->gen_key = default_gen_key;
+ klass->create_element = default_create_element;
+ klass->construct = default_construct;
+ klass->configure = default_configure;
+ klass->create_pipeline = default_create_pipeline;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmediafactory", 0,
+ "GstRTSPMediaFactory");
+}
+
+static void
+gst_rtsp_media_factory_init (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv =
+ GST_RTSP_MEDIA_FACTORY_GET_PRIVATE (factory);
+ factory->priv = priv;
+
+ priv->launch = g_strdup (DEFAULT_LAUNCH);
+ priv->shared = DEFAULT_SHARED;
+ priv->suspend_mode = DEFAULT_SUSPEND_MODE;
+ priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
+ priv->profiles = DEFAULT_PROFILES;
+ priv->protocols = DEFAULT_PROTOCOLS;
+ priv->buffer_size = DEFAULT_BUFFER_SIZE;
+
+ g_mutex_init (&priv->lock);
+ g_mutex_init (&priv->medias_lock);
+ priv->medias = g_hash_table_new_full (g_str_hash, g_str_equal,
+ g_free, g_object_unref);
+}
+
+static void
+gst_rtsp_media_factory_finalize (GObject * obj)
+{
+ GstRTSPMediaFactory *factory = GST_RTSP_MEDIA_FACTORY (obj);
+ GstRTSPMediaFactoryPrivate *priv = factory->priv;
+
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+ g_hash_table_unref (priv->medias);
+ g_mutex_clear (&priv->medias_lock);
+ g_free (priv->launch);
+ g_mutex_clear (&priv->lock);
+ if (priv->pool)
+ g_object_unref (priv->pool);
+
+ G_OBJECT_CLASS (gst_rtsp_media_factory_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_media_factory_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMediaFactory *factory = GST_RTSP_MEDIA_FACTORY (object);
+
+ switch (propid) {
+ case PROP_LAUNCH:
+ g_value_take_string (value, gst_rtsp_media_factory_get_launch (factory));
+ break;
+ case PROP_SHARED:
+ g_value_set_boolean (value, gst_rtsp_media_factory_is_shared (factory));
+ break;
+ case PROP_SUSPEND_MODE:
+ g_value_set_enum (value,
+ gst_rtsp_media_factory_get_suspend_mode (factory));
+ break;
+ case PROP_EOS_SHUTDOWN:
+ g_value_set_boolean (value,
+ gst_rtsp_media_factory_is_eos_shutdown (factory));
+ break;
+ case PROP_PROFILES:
+ g_value_set_flags (value, gst_rtsp_media_factory_get_profiles (factory));
+ break;
+ case PROP_PROTOCOLS:
+ g_value_set_flags (value, gst_rtsp_media_factory_get_protocols (factory));
+ break;
+ case PROP_BUFFER_SIZE:
+ g_value_set_uint (value,
+ gst_rtsp_media_factory_get_buffer_size (factory));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_media_factory_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMediaFactory *factory = GST_RTSP_MEDIA_FACTORY (object);
+
+ switch (propid) {
+ case PROP_LAUNCH:
+ gst_rtsp_media_factory_set_launch (factory, g_value_get_string (value));
+ break;
+ case PROP_SHARED:
+ gst_rtsp_media_factory_set_shared (factory, g_value_get_boolean (value));
+ break;
+ case PROP_SUSPEND_MODE:
+ gst_rtsp_media_factory_set_suspend_mode (factory,
+ g_value_get_enum (value));
+ break;
+ case PROP_EOS_SHUTDOWN:
+ gst_rtsp_media_factory_set_eos_shutdown (factory,
+ g_value_get_boolean (value));
+ break;
+ case PROP_PROFILES:
+ gst_rtsp_media_factory_set_profiles (factory, g_value_get_flags (value));
+ break;
+ case PROP_PROTOCOLS:
+ gst_rtsp_media_factory_set_protocols (factory, g_value_get_flags (value));
+ break;
+ case PROP_BUFFER_SIZE:
+ gst_rtsp_media_factory_set_buffer_size (factory,
+ g_value_get_uint (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_media_factory_new:
+ *
+ * Create a new #GstRTSPMediaFactory instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPMediaFactory object.
+ */
+GstRTSPMediaFactory *
+gst_rtsp_media_factory_new (void)
+{
+ GstRTSPMediaFactory *result;
+
+ result = g_object_new (GST_TYPE_RTSP_MEDIA_FACTORY, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_permissions:
+ * @factory: a #GstRTSPMediaFactory
+ * @permissions: (transfer none): a #GstRTSPPermissions
+ *
+ * Set @permissions on @factory.
+ */
+void
+gst_rtsp_media_factory_set_permissions (GstRTSPMediaFactory * factory,
+ GstRTSPPermissions * permissions)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+ if ((priv->permissions = permissions))
+ gst_rtsp_permissions_ref (permissions);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_permissions:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the permissions object from @factory.
+ *
+ * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
+ */
+GstRTSPPermissions *
+gst_rtsp_media_factory_get_permissions (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPPermissions *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if ((result = priv->permissions))
+ gst_rtsp_permissions_ref (result);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_add_role:
+ * @factory: a #GstRTSPMediaFactory
+ * @role: a role
+ * @fieldname: the first field name
+ * @...: additional arguments
+ *
+ * A convenience method to add @role with @fieldname and additional arguments to
+ * the permissions of @factory. If @factory had no permissions, new permissions
+ * will be created and the role will be added to it.
+ */
+void
+gst_rtsp_media_factory_add_role (GstRTSPMediaFactory * factory,
+ const gchar * role, const gchar * fieldname, ...)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ va_list var_args;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+ g_return_if_fail (role != NULL);
+ g_return_if_fail (fieldname != NULL);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if (priv->permissions == NULL)
+ priv->permissions = gst_rtsp_permissions_new ();
+
+ va_start (var_args, fieldname);
+ gst_rtsp_permissions_add_role_valist (priv->permissions, role, fieldname,
+ var_args);
+ va_end (var_args);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_set_launch:
+ * @factory: a #GstRTSPMediaFactory
+ * @launch: the launch description
+ *
+ *
+ * The gst_parse_launch() line to use for constructing the pipeline in the
+ * default prepare vmethod.
+ *
+ * The pipeline description should return a GstBin as the toplevel element
+ * which can be accomplished by enclosing the dscription with brackets '('
+ * ')'.
+ *
+ * The description should return a pipeline with payloaders named pay0, pay1,
+ * etc.. Each of the payloaders will result in a stream.
+ */
+void
+gst_rtsp_media_factory_set_launch (GstRTSPMediaFactory * factory,
+ const gchar * launch)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+ g_return_if_fail (launch != NULL);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ g_free (priv->launch);
+ priv->launch = g_strdup (launch);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_launch:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the gst_parse_launch() pipeline description that will be used in the
+ * default prepare vmethod.
+ *
+ * Returns: (transfer full): the configured launch description. g_free() after
+ * usage.
+ */
+gchar *
+gst_rtsp_media_factory_get_launch (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = g_strdup (priv->launch);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_suspend_mode:
+ * @factory: a #GstRTSPMediaFactory
+ * @mode: the new #GstRTSPSuspendMode
+ *
+ * Configure how media created from this factory will be suspended.
+ */
+void
+gst_rtsp_media_factory_set_suspend_mode (GstRTSPMediaFactory * factory,
+ GstRTSPSuspendMode mode)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->suspend_mode = mode;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_suspend_mode:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get how media created from this factory will be suspended.
+ *
+ * Returns: a #GstRTSPSuspendMode.
+ */
+GstRTSPSuspendMode
+gst_rtsp_media_factory_get_suspend_mode (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPSuspendMode result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory),
+ GST_RTSP_SUSPEND_MODE_NONE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->suspend_mode;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_shared:
+ * @factory: a #GstRTSPMediaFactory
+ * @shared: the new value
+ *
+ * Configure if media created from this factory can be shared between clients.
+ */
+void
+gst_rtsp_media_factory_set_shared (GstRTSPMediaFactory * factory,
+ gboolean shared)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->shared = shared;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_is_shared:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get if media created from this factory can be shared between clients.
+ *
+ * Returns: %TRUE if the media will be shared between clients.
+ */
+gboolean
+gst_rtsp_media_factory_is_shared (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->shared;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_eos_shutdown:
+ * @factory: a #GstRTSPMediaFactory
+ * @eos_shutdown: the new value
+ *
+ * Configure if media created from this factory will have an EOS sent to the
+ * pipeline before shutdown.
+ */
+void
+gst_rtsp_media_factory_set_eos_shutdown (GstRTSPMediaFactory * factory,
+ gboolean eos_shutdown)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->eos_shutdown = eos_shutdown;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_is_eos_shutdown:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get if media created from this factory will have an EOS event sent to the
+ * pipeline before shutdown.
+ *
+ * Returns: %TRUE if the media will receive EOS before shutdown.
+ */
+gboolean
+gst_rtsp_media_factory_is_eos_shutdown (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->eos_shutdown;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_buffer_size:
+ * @factory: a #GstRTSPMedia
+ * @size: the new value
+ *
+ * Set the kernel UDP buffer size.
+ */
+void
+gst_rtsp_media_factory_set_buffer_size (GstRTSPMediaFactory * factory,
+ guint size)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->buffer_size = size;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_buffer_size:
+ * @factory: a #GstRTSPMedia
+ *
+ * Get the kernel UDP buffer size.
+ *
+ * Returns: the kernel UDP buffer size.
+ */
+guint
+gst_rtsp_media_factory_get_buffer_size (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ guint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), 0);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->buffer_size;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_address_pool:
+ * @factory: a #GstRTSPMediaFactory
+ * @pool: (transfer none): a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @factory.
+ */
+void
+gst_rtsp_media_factory_set_address_pool (GstRTSPMediaFactory * factory,
+ GstRTSPAddressPool * pool)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPAddressPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if ((old = priv->pool) != pool)
+ priv->pool = pool ? g_object_ref (pool) : NULL;
+ else
+ old = NULL;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_media_factory_get_address_pool:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @factory.
+ *
+ * Returns: (transfer full): the #GstRTSPAddressPool of @factory. g_object_unref() after
+ * usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_media_factory_get_address_pool (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPAddressPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if ((result = priv->pool))
+ g_object_ref (result);
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_profiles:
+ * @factory: a #GstRTSPMediaFactory
+ * @profiles: the new flags
+ *
+ * Configure the allowed profiles for @factory.
+ */
+void
+gst_rtsp_media_factory_set_profiles (GstRTSPMediaFactory * factory,
+ GstRTSPProfile profiles)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_DEBUG_OBJECT (factory, "profiles %d", profiles);
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->profiles = profiles;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_profiles:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the allowed profiles of @factory.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_media_factory_get_profiles (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory),
+ GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ res = priv->profiles;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_factory_set_protocols:
+ * @factory: a #GstRTSPMediaFactory
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @factory.
+ */
+void
+gst_rtsp_media_factory_set_protocols (GstRTSPMediaFactory * factory,
+ GstRTSPLowerTrans protocols)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+ priv = factory->priv;
+
+ GST_DEBUG_OBJECT (factory, "protocols %d", protocols);
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ priv->protocols = protocols;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_protocols:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the allowed protocols of @factory.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_media_factory_get_protocols (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ GstRTSPLowerTrans res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory),
+ GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ res = priv->protocols;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return res;
+}
+
+static gboolean
+compare_media (gpointer key, GstRTSPMedia * media1, GstRTSPMedia * media2)
+{
+ return (media1 == media2);
+}
+
+static void
+media_unprepared (GstRTSPMedia * media, GWeakRef * ref)
+{
+ GstRTSPMediaFactory *factory = g_weak_ref_get (ref);
+ GstRTSPMediaFactoryPrivate *priv;
+
+ if (!factory)
+ return;
+
+ priv = factory->priv;;
+
+ g_mutex_lock (&priv->medias_lock);
+ g_hash_table_foreach_remove (priv->medias, (GHRFunc) compare_media, media);
+ g_mutex_unlock (&priv->medias_lock);
+
+ g_object_unref (factory);
+}
+
+static GWeakRef *
+weak_ref_new (gpointer obj)
+{
+ GWeakRef *ref = g_slice_new (GWeakRef);
+
+ g_weak_ref_init (ref, obj);
+ return ref;
+}
+
+static void
+weak_ref_free (GWeakRef * ref)
+{
+ g_weak_ref_clear (ref);
+ g_slice_free (GWeakRef, ref);
+}
+
+/**
+ * gst_rtsp_media_factory_construct:
+ * @factory: a #GstRTSPMediaFactory
+ * @url: the url used
+ *
+ * Construct the media object and create its streams. Implementations
+ * should create the needed gstreamer elements and add them to the result
+ * object. No state changes should be performed on them yet.
+ *
+ * One or more GstRTSPStream objects should be created from the result
+ * with gst_rtsp_media_create_stream ().
+ *
+ * After the media is constructed, it can be configured and then prepared
+ * with gst_rtsp_media_prepare ().
+ *
+ * Returns: (transfer full): a new #GstRTSPMedia if the media could be prepared.
+ */
+GstRTSPMedia *
+gst_rtsp_media_factory_construct (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ gchar *key;
+ GstRTSPMedia *media;
+ GstRTSPMediaFactoryClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+ g_return_val_if_fail (url != NULL, NULL);
+
+ priv = factory->priv;;
+ klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+ /* convert the url to a key for the hashtable. NULL return or a NULL function
+ * will not cache anything for this factory. */
+ if (klass->gen_key)
+ key = klass->gen_key (factory, url);
+ else
+ key = NULL;
+
+ g_mutex_lock (&priv->medias_lock);
+ if (key) {
+ /* we have a key, see if we find a cached media */
+ media = g_hash_table_lookup (priv->medias, key);
+ if (media)
+ g_object_ref (media);
+ } else
+ media = NULL;
+
+ if (media == NULL) {
+ /* nothing cached found, try to create one */
+ if (klass->construct) {
+ media = klass->construct (factory, url);
+ if (media)
+ g_signal_emit (factory,
+ gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONSTRUCTED], 0, media,
+ NULL);
+ } else
+ media = NULL;
+
+ if (media) {
+ /* configure the media */
+ if (klass->configure)
+ klass->configure (factory, media);
+
+ g_signal_emit (factory,
+ gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONFIGURE], 0, media,
+ NULL);
+
+ /* check if we can cache this media */
+ if (gst_rtsp_media_is_shared (media)) {
+ /* insert in the hashtable, takes ownership of the key */
+ g_object_ref (media);
+ g_hash_table_insert (priv->medias, key, media);
+ key = NULL;
+ }
+ if (!gst_rtsp_media_is_reusable (media)) {
+ /* when not reusable, connect to the unprepare signal to remove the item
+ * from our cache when it gets unprepared */
+ g_signal_connect_data (media, "unprepared",
+ (GCallback) media_unprepared, weak_ref_new (factory),
+ (GClosureNotify) weak_ref_free, 0);
+ }
+ }
+ }
+ g_mutex_unlock (&priv->medias_lock);
+
+ if (key)
+ g_free (key);
+
+ GST_INFO ("constructed media %p for url %s", media, url->abspath);
+
+ return media;
+}
+
+static gchar *
+default_gen_key (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
+{
+ gchar *result;
+ const gchar *pre_query;
+ const gchar *query;
+
+ pre_query = url->query ? "?" : "";
+ query = url->query ? url->query : "";
+
+ result =
+ g_strdup_printf ("%u%s%s%s", url->port, url->abspath, pre_query, query);
+
+ return result;
+}
+
+static GstElement *
+default_create_element (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
+{
+ GstRTSPMediaFactoryPrivate *priv = factory->priv;
+ GstElement *element;
+ GError *error = NULL;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ /* we need a parse syntax */
+ if (priv->launch == NULL)
+ goto no_launch;
+
+ /* parse the user provided launch line */
+ element = gst_parse_launch (priv->launch, &error);
+ if (element == NULL)
+ goto parse_error;
+
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ if (error != NULL) {
+ /* a recoverable error was encountered */
+ GST_WARNING ("recoverable parsing error: %s", error->message);
+ g_error_free (error);
+ }
+ return element;
+
+ /* ERRORS */
+no_launch:
+ {
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+ g_critical ("no launch line specified");
+ return NULL;
+ }
+parse_error:
+ {
+ g_critical ("could not parse launch syntax (%s): %s", priv->launch,
+ (error ? error->message : "unknown reason"));
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+ if (error)
+ g_error_free (error);
+ return NULL;
+ }
+}
+
+static GstRTSPMedia *
+default_construct (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
+{
+ GstRTSPMedia *media;
+ GstElement *element, *pipeline;
+ GstRTSPMediaFactoryClass *klass;
+
+ klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+ if (!klass->create_pipeline)
+ goto no_create;
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ if (element == NULL)
+ goto no_element;
+
+ /* create a new empty media */
+ media = gst_rtsp_media_new (element);
+
+ gst_rtsp_media_collect_streams (media);
+
+ pipeline = klass->create_pipeline (factory, media);
+ if (pipeline == NULL)
+ goto no_pipeline;
+
+ return media;
+
+ /* ERRORS */
+no_create:
+ {
+ g_critical ("no create_pipeline function");
+ return NULL;
+ }
+no_element:
+ {
+ g_critical ("could not create element");
+ return NULL;
+ }
+no_pipeline:
+ {
+ g_critical ("can't create pipeline");
+ g_object_unref (media);
+ return NULL;
+ }
+}
+
+static GstElement *
+default_create_pipeline (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+ GstElement *pipeline;
+
+ pipeline = gst_pipeline_new ("media-pipeline");
+ gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+ return pipeline;
+}
+
+static void
+default_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+ GstRTSPMediaFactoryPrivate *priv = factory->priv;
+ gboolean shared, eos_shutdown;
+ guint size;
+ GstRTSPSuspendMode suspend_mode;
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans protocols;
+ GstRTSPAddressPool *pool;
+ GstRTSPPermissions *perms;
+
+ /* configure the sharedness */
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ suspend_mode = priv->suspend_mode;
+ shared = priv->shared;
+ eos_shutdown = priv->eos_shutdown;
+ size = priv->buffer_size;
+ profiles = priv->profiles;
+ protocols = priv->protocols;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ gst_rtsp_media_set_suspend_mode (media, suspend_mode);
+ gst_rtsp_media_set_shared (media, shared);
+ gst_rtsp_media_set_eos_shutdown (media, eos_shutdown);
+ gst_rtsp_media_set_buffer_size (media, size);
+ gst_rtsp_media_set_profiles (media, profiles);
+ gst_rtsp_media_set_protocols (media, protocols);
+
+ if ((pool = gst_rtsp_media_factory_get_address_pool (factory))) {
+ gst_rtsp_media_set_address_pool (media, pool);
+ g_object_unref (pool);
+ }
+ if ((perms = gst_rtsp_media_factory_get_permissions (factory))) {
+ gst_rtsp_media_set_permissions (media, perms);
+ gst_rtsp_permissions_unref (perms);
+ }
+}
+
+/**
+ * gst_rtsp_media_factory_create_element:
+ * @factory: a #GstRTSPMediaFactory
+ * @url: the url used
+ *
+ * Construct and return a #GstElement that is a #GstBin containing
+ * the elements to use for streaming the media.
+ *
+ * The bin should contain payloaders pay\%d for each stream. The default
+ * implementation of this function returns the bin created from the
+ * launch parameter.
+ *
+ * Returns: (transfer floating): a new #GstElement.
+ */
+GstElement *
+gst_rtsp_media_factory_create_element (GstRTSPMediaFactory * factory,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMediaFactoryClass *klass;
+ GstElement *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+ g_return_val_if_fail (url != NULL, NULL);
+
+ klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+ if (klass->create_element)
+ result = klass->create_element (factory, url);
+ else
+ result = NULL;
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspurl.h>
+
+#include "rtsp-media.h"
+#include "rtsp-permissions.h"
+#include "rtsp-address-pool.h"
+
+#ifndef __GST_RTSP_MEDIA_FACTORY_H__
+#define __GST_RTSP_MEDIA_FACTORY_H__
+
+G_BEGIN_DECLS
+
+/* types for the media factory */
+#define GST_TYPE_RTSP_MEDIA_FACTORY (gst_rtsp_media_factory_get_type ())
+#define GST_IS_RTSP_MEDIA_FACTORY(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY))
+#define GST_IS_RTSP_MEDIA_FACTORY_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA_FACTORY))
+#define GST_RTSP_MEDIA_FACTORY_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactoryClass))
+#define GST_RTSP_MEDIA_FACTORY(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactory))
+#define GST_RTSP_MEDIA_FACTORY_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactoryClass))
+#define GST_RTSP_MEDIA_FACTORY_CAST(obj) ((GstRTSPMediaFactory*)(obj))
+#define GST_RTSP_MEDIA_FACTORY_CLASS_CAST(klass) ((GstRTSPMediaFactoryClass*)(klass))
+
+typedef struct _GstRTSPMediaFactory GstRTSPMediaFactory;
+typedef struct _GstRTSPMediaFactoryClass GstRTSPMediaFactoryClass;
+typedef struct _GstRTSPMediaFactoryPrivate GstRTSPMediaFactoryPrivate;
+
+/**
+ * GstRTSPMediaFactory:
+ *
+ * The definition and logic for constructing the pipeline for a media. The media
+ * can contain multiple streams like audio and video.
+ */
+struct _GstRTSPMediaFactory {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPMediaFactoryPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaFactoryClass:
+ * @gen_key: convert @url to a key for caching shared #GstRTSPMedia objects.
+ * The default implementation of this function will use the complete URL
+ * including the query parameters to return a key.
+ * @create_element: Construct and return a #GstElement that is a #GstBin containing
+ * the elements to use for streaming the media. The bin should contain
+ * payloaders pay\%d for each stream. The default implementation of this
+ * function returns the bin created from the launch parameter.
+ * @construct: the vmethod that will be called when the factory has to create the
+ * #GstRTSPMedia for @url. The default implementation of this
+ * function calls create_element to retrieve an element and then looks for
+ * pay\%d to create the streams.
+ * @create_pipeline: create a new pipeline or re-use an existing one and
+ * add the #GstRTSPMedia's element created by @construct to the pipeline.
+ * @configure: configure the media created with @construct. The default
+ * implementation will configure the 'shared' property of the media.
+ * @media_constructed: signal emited when a media was constructed
+ * @media_configure: signal emited when a media should be configured
+ *
+ * The #GstRTSPMediaFactory class structure.
+ */
+struct _GstRTSPMediaFactoryClass {
+ GObjectClass parent_class;
+
+ gchar * (*gen_key) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+
+ GstElement * (*create_element) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+ GstRTSPMedia * (*construct) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+ GstElement * (*create_pipeline) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+ void (*configure) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+
+ /* signals */
+ void (*media_constructed) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+ void (*media_configure) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType gst_rtsp_media_factory_get_type (void);
+
+/* creating the factory */
+GstRTSPMediaFactory * gst_rtsp_media_factory_new (void);
+
+/* configuring the factory */
+void gst_rtsp_media_factory_set_launch (GstRTSPMediaFactory *factory,
+ const gchar *launch);
+gchar * gst_rtsp_media_factory_get_launch (GstRTSPMediaFactory *factory);
+
+void gst_rtsp_media_factory_set_permissions (GstRTSPMediaFactory *factory,
+ GstRTSPPermissions *permissions);
+GstRTSPPermissions * gst_rtsp_media_factory_get_permissions (GstRTSPMediaFactory *factory);
+void gst_rtsp_media_factory_add_role (GstRTSPMediaFactory *factory,
+ const gchar *role,
+ const gchar *fieldname, ...);
+
+void gst_rtsp_media_factory_set_shared (GstRTSPMediaFactory *factory,
+ gboolean shared);
+gboolean gst_rtsp_media_factory_is_shared (GstRTSPMediaFactory *factory);
+
+void gst_rtsp_media_factory_set_suspend_mode (GstRTSPMediaFactory *factory,
+ GstRTSPSuspendMode mode);
+GstRTSPSuspendMode gst_rtsp_media_factory_get_suspend_mode (GstRTSPMediaFactory *factory);
+
+void gst_rtsp_media_factory_set_eos_shutdown (GstRTSPMediaFactory *factory,
+ gboolean eos_shutdown);
+gboolean gst_rtsp_media_factory_is_eos_shutdown (GstRTSPMediaFactory *factory);
+
+void gst_rtsp_media_factory_set_profiles (GstRTSPMediaFactory *factory,
+ GstRTSPProfile profiles);
+GstRTSPProfile gst_rtsp_media_factory_get_profiles (GstRTSPMediaFactory *factory);
+
+void gst_rtsp_media_factory_set_protocols (GstRTSPMediaFactory *factory,
+ GstRTSPLowerTrans protocols);
+GstRTSPLowerTrans gst_rtsp_media_factory_get_protocols (GstRTSPMediaFactory *factory);
+
+void gst_rtsp_media_factory_set_address_pool (GstRTSPMediaFactory * factory,
+ GstRTSPAddressPool * pool);
+GstRTSPAddressPool * gst_rtsp_media_factory_get_address_pool (GstRTSPMediaFactory * factory);
+
+void gst_rtsp_media_factory_set_buffer_size (GstRTSPMediaFactory * factory,
+ guint size);
+guint gst_rtsp_media_factory_get_buffer_size (GstRTSPMediaFactory * factory);
+
+/* creating the media from the factory and a url */
+GstRTSPMedia * gst_rtsp_media_factory_construct (GstRTSPMediaFactory *factory,
+ const GstRTSPUrl *url);
+
+GstElement * gst_rtsp_media_factory_create_element (GstRTSPMediaFactory *factory,
+ const GstRTSPUrl *url);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MEDIA_FACTORY_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media
+ * @short_description: The media pipeline
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
+ * #GstRTSPSessionMedia
+ *
+ * a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
+ * streaming to the clients. The actual data transfer is done by the
+ * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
+ *
+ * The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
+ * client does a DESCRIBE or SETUP of a resource.
+ *
+ * A media is created with gst_rtsp_media_new() that takes the element that will
+ * provide the streaming elements. For each of the streams, a new #GstRTSPStream
+ * object needs to be made with the gst_rtsp_media_create_stream() which takes
+ * the payloader element and the source pad that produces the RTP stream.
+ *
+ * The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
+ * prepare method will add rtpbin and sinks and sources to send and receive RTP
+ * and RTCP packets from the clients. Each stream srcpad is connected to an
+ * input into the internal rtpbin.
+ *
+ * It is also possible to dynamically create #GstRTSPStream objects during the
+ * prepare phase. With gst_rtsp_media_get_status() you can check the status of
+ * the prepare phase.
+ *
+ * After the media is prepared, it is ready for streaming. It will usually be
+ * managed in a session with gst_rtsp_session_manage_media(). See
+ * #GstRTSPSession and #GstRTSPSessionMedia.
+ *
+ * The state of the media can be controlled with gst_rtsp_media_set_state ().
+ * Seeking can be done with gst_rtsp_media_seek().
+ *
+ * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
+ * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
+ * cleanly shut down.
+ *
+ * With gst_rtsp_media_set_shared(), the media can be shared between multiple
+ * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
+ * can be prepared again after an unprepare.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include <string.h>
+#include <stdlib.h>
+
+#include <gst/app/gstappsrc.h>
+#include <gst/app/gstappsink.h>
+
+#include "rtsp-media.h"
+
+#define GST_RTSP_MEDIA_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaPrivate))
+
+struct _GstRTSPMediaPrivate
+{
+ GMutex lock;
+ GCond cond;
+
+ /* protected by lock */
+ GstRTSPPermissions *permissions;
+ gboolean shared;
+ gboolean suspend_mode;
+ gboolean reusable;
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans protocols;
+ gboolean reused;
+ gboolean eos_shutdown;
+ guint buffer_size;
+ GstRTSPAddressPool *pool;
+ gboolean blocked;
+
+ GstElement *element;
+ GRecMutex state_lock; /* locking order: state lock, lock */
+ GPtrArray *streams; /* protected by lock */
+ GList *dynamic; /* protected by lock */
+ GstRTSPMediaStatus status; /* protected by lock */
+ gint prepare_count;
+ gint n_active;
+ gboolean adding;
+
+ /* the pipeline for the media */
+ GstElement *pipeline;
+ GstElement *fakesink; /* protected by lock */
+ GSource *source;
+ guint id;
+ GstRTSPThread *thread;
+
+ gboolean time_provider;
+ GstNetTimeProvider *nettime;
+
+ gboolean is_live;
+ gboolean seekable;
+ gboolean buffering;
+ GstState target_state;
+
+ /* RTP session manager */
+ GstElement *rtpbin;
+
+ /* the range of media */
+ GstRTSPTimeRange range; /* protected by lock */
+ GstClockTime range_start;
+ GstClockTime range_stop;
+};
+
+#define DEFAULT_SHARED FALSE
+#define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
+#define DEFAULT_REUSABLE FALSE
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+ GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_EOS_SHUTDOWN FALSE
+#define DEFAULT_BUFFER_SIZE 0x80000
+#define DEFAULT_TIME_PROVIDER FALSE
+
+/* define to dump received RTCP packets */
+#undef DUMP_STATS
+
+enum
+{
+ PROP_0,
+ PROP_SHARED,
+ PROP_SUSPEND_MODE,
+ PROP_REUSABLE,
+ PROP_PROFILES,
+ PROP_PROTOCOLS,
+ PROP_EOS_SHUTDOWN,
+ PROP_BUFFER_SIZE,
+ PROP_ELEMENT,
+ PROP_TIME_PROVIDER,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_NEW_STREAM,
+ SIGNAL_REMOVED_STREAM,
+ SIGNAL_PREPARED,
+ SIGNAL_UNPREPARED,
+ SIGNAL_TARGET_STATE,
+ SIGNAL_NEW_STATE,
+ SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
+#define GST_CAT_DEFAULT rtsp_media_debug
+
+static void gst_rtsp_media_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_finalize (GObject * obj);
+
+static gboolean default_handle_message (GstRTSPMedia * media,
+ GstMessage * message);
+static void finish_unprepare (GstRTSPMedia * media);
+static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
+static gboolean default_unprepare (GstRTSPMedia * media);
+static gboolean default_suspend (GstRTSPMedia * media);
+static gboolean default_unsuspend (GstRTSPMedia * media);
+static gboolean default_convert_range (GstRTSPMedia * media,
+ GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
+static gboolean default_query_position (GstRTSPMedia * media,
+ gint64 * position);
+static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
+static GstElement *default_create_rtpbin (GstRTSPMedia * media);
+static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info);
+
+static gboolean wait_preroll (GstRTSPMedia * media);
+
+static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
+
+#define C_ENUM(v) ((gint) v)
+
+#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
+GType
+gst_rtsp_suspend_mode_get_type (void)
+{
+ static gsize id = 0;
+ static const GEnumValue values[] = {
+ {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
+ {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
+ "pause"},
+ {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
+ "reset"},
+ {0, NULL, NULL}
+ };
+
+ if (g_once_init_enter (&id)) {
+ GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
+ g_once_init_leave (&id, tmp);
+ }
+ return (GType) id;
+}
+
+G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_media_get_property;
+ gobject_class->set_property = gst_rtsp_media_set_property;
+ gobject_class->finalize = gst_rtsp_media_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_SHARED,
+ g_param_spec_boolean ("shared", "Shared",
+ "If this media pipeline can be shared", DEFAULT_SHARED,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
+ g_param_spec_enum ("suspend-mode", "Suspend Mode",
+ "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
+ DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_REUSABLE,
+ g_param_spec_boolean ("reusable", "Reusable",
+ "If this media pipeline can be reused after an unprepare",
+ DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
+ g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
+ "Send an EOS event to the pipeline before unpreparing",
+ DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
+ g_param_spec_uint ("buffer-size", "Buffer Size",
+ "The kernel UDP buffer size to use", 0, G_MAXUINT,
+ DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_ELEMENT,
+ g_param_spec_object ("element", "The Element",
+ "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
+ G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
+ g_param_spec_boolean ("time-provider", "Time Provider",
+ "Use a NetTimeProvider for clients",
+ DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
+ g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
+ g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
+
+ gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
+ g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_STREAM);
+
+ gst_rtsp_media_signals[SIGNAL_PREPARED] =
+ g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
+ g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+ gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
+ g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
+ g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+ gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
+ g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
+
+ gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
+ g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
+ g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
+
+ klass->handle_message = default_handle_message;
+ klass->prepare = default_prepare;
+ klass->unprepare = default_unprepare;
+ klass->suspend = default_suspend;
+ klass->unsuspend = default_unsuspend;
+ klass->convert_range = default_convert_range;
+ klass->query_position = default_query_position;
+ klass->query_stop = default_query_stop;
+ klass->create_rtpbin = default_create_rtpbin;
+ klass->setup_sdp = default_setup_sdp;
+}
+
+static void
+gst_rtsp_media_init (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
+
+ media->priv = priv;
+
+ priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
+ g_mutex_init (&priv->lock);
+ g_cond_init (&priv->cond);
+ g_rec_mutex_init (&priv->state_lock);
+
+ priv->shared = DEFAULT_SHARED;
+ priv->suspend_mode = DEFAULT_SUSPEND_MODE;
+ priv->reusable = DEFAULT_REUSABLE;
+ priv->profiles = DEFAULT_PROFILES;
+ priv->protocols = DEFAULT_PROTOCOLS;
+ priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
+ priv->buffer_size = DEFAULT_BUFFER_SIZE;
+ priv->time_provider = DEFAULT_TIME_PROVIDER;
+}
+
+static void
+gst_rtsp_media_finalize (GObject * obj)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMedia *media;
+
+ media = GST_RTSP_MEDIA (obj);
+ priv = media->priv;
+
+ GST_INFO ("finalize media %p", media);
+
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+
+ g_ptr_array_unref (priv->streams);
+
+ g_list_free_full (priv->dynamic, gst_object_unref);
+
+ if (priv->pipeline)
+ gst_object_unref (priv->pipeline);
+ if (priv->nettime)
+ gst_object_unref (priv->nettime);
+ gst_object_unref (priv->element);
+ if (priv->pool)
+ g_object_unref (priv->pool);
+ g_mutex_clear (&priv->lock);
+ g_cond_clear (&priv->cond);
+ g_rec_mutex_clear (&priv->state_lock);
+
+ G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_media_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMedia *media = GST_RTSP_MEDIA (object);
+
+ switch (propid) {
+ case PROP_ELEMENT:
+ g_value_set_object (value, media->priv->element);
+ break;
+ case PROP_SHARED:
+ g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
+ break;
+ case PROP_SUSPEND_MODE:
+ g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
+ break;
+ case PROP_REUSABLE:
+ g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
+ break;
+ case PROP_PROFILES:
+ g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
+ break;
+ case PROP_PROTOCOLS:
+ g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
+ break;
+ case PROP_EOS_SHUTDOWN:
+ g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
+ break;
+ case PROP_BUFFER_SIZE:
+ g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
+ break;
+ case PROP_TIME_PROVIDER:
+ g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_media_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPMedia *media = GST_RTSP_MEDIA (object);
+
+ switch (propid) {
+ case PROP_ELEMENT:
+ media->priv->element = g_value_get_object (value);
+ gst_object_ref_sink (media->priv->element);
+ break;
+ case PROP_SHARED:
+ gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
+ break;
+ case PROP_SUSPEND_MODE:
+ gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
+ break;
+ case PROP_REUSABLE:
+ gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
+ break;
+ case PROP_PROFILES:
+ gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
+ break;
+ case PROP_PROTOCOLS:
+ gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
+ break;
+ case PROP_EOS_SHUTDOWN:
+ gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
+ break;
+ case PROP_BUFFER_SIZE:
+ gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
+ break;
+ case PROP_TIME_PROVIDER:
+ gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+typedef struct
+{
+ gint64 position;
+ gboolean ret;
+} DoQueryPositionData;
+
+static void
+do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
+{
+ gint64 tmp;
+
+ if (gst_rtsp_stream_query_position (stream, &tmp)) {
+ data->position = MAX (data->position, tmp);
+ data->ret = TRUE;
+ }
+}
+
+static gboolean
+default_query_position (GstRTSPMedia * media, gint64 * position)
+{
+ GstRTSPMediaPrivate *priv;
+ DoQueryPositionData data;
+
+ priv = media->priv;
+
+ data.position = -1;
+ data.ret = FALSE;
+
+ g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
+
+ *position = data.position;
+
+ return data.ret;
+}
+
+typedef struct
+{
+ gint64 stop;
+ gboolean ret;
+} DoQueryStopData;
+
+static void
+do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
+{
+ gint64 tmp;
+
+ if (gst_rtsp_stream_query_stop (stream, &tmp)) {
+ data->stop = MAX (data->stop, tmp);
+ data->ret = TRUE;
+ }
+}
+
+static gboolean
+default_query_stop (GstRTSPMedia * media, gint64 * stop)
+{
+ GstRTSPMediaPrivate *priv;
+ DoQueryStopData data;
+
+ priv = media->priv;
+
+ data.stop = -1;
+ data.ret = FALSE;
+
+ g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
+
+ *stop = data.stop;
+
+ return data.ret;
+}
+
+static GstElement *
+default_create_rtpbin (GstRTSPMedia * media)
+{
+ GstElement *rtpbin;
+
+ rtpbin = gst_element_factory_make ("rtpbin", NULL);
+
+ return rtpbin;
+}
+
+/* must be called with state lock */
+static void
+collect_media_stats (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gint64 position = 0, stop = -1;
+
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+ priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
+ return;
+
+ priv->range.unit = GST_RTSP_RANGE_NPT;
+
+ GST_INFO ("collect media stats");
+
+ if (priv->is_live) {
+ priv->range.min.type = GST_RTSP_TIME_NOW;
+ priv->range.min.seconds = -1;
+ priv->range_start = -1;
+ priv->range.max.type = GST_RTSP_TIME_END;
+ priv->range.max.seconds = -1;
+ priv->range_stop = -1;
+ } else {
+ GstRTSPMediaClass *klass;
+ gboolean ret;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ /* get the position */
+ ret = FALSE;
+ if (klass->query_position)
+ ret = klass->query_position (media, &position);
+
+ if (!ret) {
+ GST_INFO ("position query failed");
+ position = 0;
+ }
+
+ /* get the current segment stop */
+ ret = FALSE;
+ if (klass->query_stop)
+ ret = klass->query_stop (media, &stop);
+
+ if (!ret) {
+ GST_INFO ("stop query failed");
+ stop = -1;
+ }
+
+ GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
+
+ if (position == -1) {
+ priv->range.min.type = GST_RTSP_TIME_NOW;
+ priv->range.min.seconds = -1;
+ priv->range_start = -1;
+ } else {
+ priv->range.min.type = GST_RTSP_TIME_SECONDS;
+ priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
+ priv->range_start = position;
+ }
+ if (stop == -1) {
+ priv->range.max.type = GST_RTSP_TIME_END;
+ priv->range.max.seconds = -1;
+ priv->range_stop = -1;
+ } else {
+ priv->range.max.type = GST_RTSP_TIME_SECONDS;
+ priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
+ priv->range_stop = stop;
+ }
+ }
+}
+
+/**
+ * gst_rtsp_media_new:
+ * @element: (transfer full): a #GstElement
+ *
+ * Create a new #GstRTSPMedia instance. @element is the bin element that
+ * provides the different streams. The #GstRTSPMedia object contains the
+ * element to produce RTP data for one or more related (audio/video/..)
+ * streams.
+ *
+ * Ownership is taken of @element.
+ *
+ * Returns: (transfer full): a new #GstRTSPMedia object.
+ */
+GstRTSPMedia *
+gst_rtsp_media_new (GstElement * element)
+{
+ GstRTSPMedia *result;
+
+ g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
+
+ result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_get_element:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the element that was used when constructing @media.
+ *
+ * Returns: (transfer full): a #GstElement. Unref after usage.
+ */
+GstElement *
+gst_rtsp_media_get_element (GstRTSPMedia * media)
+{
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ return gst_object_ref (media->priv->element);
+}
+
+/**
+ * gst_rtsp_media_take_pipeline:
+ * @media: a #GstRTSPMedia
+ * @pipeline: (transfer full): a #GstPipeline
+ *
+ * Set @pipeline as the #GstPipeline for @media. Ownership is
+ * taken of @pipeline.
+ */
+void
+gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
+{
+ GstRTSPMediaPrivate *priv;
+ GstElement *old;
+ GstNetTimeProvider *nettime;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+ g_return_if_fail (GST_IS_PIPELINE (pipeline));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ old = priv->pipeline;
+ priv->pipeline = GST_ELEMENT_CAST (pipeline);
+ nettime = priv->nettime;
+ priv->nettime = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ gst_object_unref (old);
+
+ if (nettime)
+ gst_object_unref (nettime);
+
+ gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
+}
+
+/**
+ * gst_rtsp_media_set_permissions:
+ * @media: a #GstRTSPMedia
+ * @permissions: (transfer none): a #GstRTSPPermissions
+ *
+ * Set @permissions on @media.
+ */
+void
+gst_rtsp_media_set_permissions (GstRTSPMedia * media,
+ GstRTSPPermissions * permissions)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+ if ((priv->permissions = permissions))
+ gst_rtsp_permissions_ref (permissions);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_permissions:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the permissions object from @media.
+ *
+ * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
+ */
+GstRTSPPermissions *
+gst_rtsp_media_get_permissions (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPPermissions *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->permissions))
+ gst_rtsp_permissions_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_set_suspend_mode:
+ * @media: a #GstRTSPMedia
+ * @mode: the new #GstRTSPSuspendMode
+ *
+ * Control how @ media will be suspended after the SDP has been generated and
+ * after a PAUSE request has been performed.
+ *
+ * Media must be unprepared when setting the suspend mode.
+ */
+void
+gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto was_prepared;
+ priv->suspend_mode = mode;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return;
+
+ /* ERRORS */
+was_prepared:
+ {
+ GST_WARNING ("media %p was prepared", media);
+ g_rec_mutex_unlock (&priv->state_lock);
+ }
+}
+
+/**
+ * gst_rtsp_media_get_suspend_mode:
+ * @media: a #GstRTSPMedia
+ *
+ * Get how @media will be suspended.
+ *
+ * Returns: #GstRTSPSuspendMode.
+ */
+GstRTSPSuspendMode
+gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPSuspendMode res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ res = priv->suspend_mode;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_shared:
+ * @media: a #GstRTSPMedia
+ * @shared: the new value
+ *
+ * Set or unset if the pipeline for @media can be shared will multiple clients.
+ * When @shared is %TRUE, client requests for this media will share the media
+ * pipeline.
+ */
+void
+gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->shared = shared;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_shared:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media can be shared between multiple clients.
+ *
+ * Returns: %TRUE if the media can be shared between clients.
+ */
+gboolean
+gst_rtsp_media_is_shared (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->shared;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_reusable:
+ * @media: a #GstRTSPMedia
+ * @reusable: the new value
+ *
+ * Set or unset if the pipeline for @media can be reused after the pipeline has
+ * been unprepared.
+ */
+void
+gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->reusable = reusable;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_reusable:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media can be reused after an unprepare.
+ *
+ * Returns: %TRUE if the media can be reused
+ */
+gboolean
+gst_rtsp_media_is_reusable (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->reusable;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+static void
+do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
+{
+ gst_rtsp_stream_set_profiles (stream, *profiles);
+}
+
+/**
+ * gst_rtsp_media_set_profiles:
+ * @media: a #GstRTSPMedia
+ * @profiles: the new flags
+ *
+ * Configure the allowed lower transport for @media.
+ */
+void
+gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->profiles = profiles;
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_profiles:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the allowed profiles of @media.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_media_get_profiles (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->profiles;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+static void
+do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
+{
+ gst_rtsp_stream_set_protocols (stream, *protocols);
+}
+
+/**
+ * gst_rtsp_media_set_protocols:
+ * @media: a #GstRTSPMedia
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @media.
+ */
+void
+gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->protocols = protocols;
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_protocols:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the allowed protocols of @media.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_media_get_protocols (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPLowerTrans res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
+ GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->protocols;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_eos_shutdown:
+ * @media: a #GstRTSPMedia
+ * @eos_shutdown: the new value
+ *
+ * Set or unset if an EOS event will be sent to the pipeline for @media before
+ * it is unprepared.
+ */
+void
+gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->eos_shutdown = eos_shutdown;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_eos_shutdown:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media will send an EOS down the pipeline before
+ * unpreparing.
+ *
+ * Returns: %TRUE if the media will send EOS before unpreparing.
+ */
+gboolean
+gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->eos_shutdown;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_buffer_size:
+ * @media: a #GstRTSPMedia
+ * @size: the new value
+ *
+ * Set the kernel UDP buffer size.
+ */
+void
+gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ GST_LOG_OBJECT (media, "set buffer size %u", size);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->buffer_size = size;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_buffer_size:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the kernel UDP buffer size.
+ *
+ * Returns: the kernel UDP buffer size.
+ */
+guint
+gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_unlock (&priv->lock);
+ res = priv->buffer_size;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_use_time_provider:
+ * @media: a #GstRTSPMedia
+ * @time_provider: if a #GstNetTimeProvider should be used
+ *
+ * Set @media to provide a #GstNetTimeProvider.
+ */
+void
+gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->time_provider = time_provider;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_time_provider:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
+ *
+ * Use gst_rtsp_media_get_time_provider() to get the network clock.
+ *
+ * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
+ */
+gboolean
+gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_unlock (&priv->lock);
+ res = priv->time_provider;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_address_pool:
+ * @media: a #GstRTSPMedia
+ * @pool: (transfer none): a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @media.
+ */
+void
+gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
+ GstRTSPAddressPool * pool)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPAddressPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ GST_LOG_OBJECT (media, "set address pool %p", pool);
+
+ g_mutex_lock (&priv->lock);
+ if ((old = priv->pool) != pool)
+ priv->pool = pool ? g_object_ref (pool) : NULL;
+ else
+ old = NULL;
+ g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
+ pool);
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_media_get_address_pool:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @media.
+ *
+ * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
+ * usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPAddressPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->pool))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_collect_streams:
+ * @media: a #GstRTSPMedia
+ *
+ * Find all payloader elements, they should be named pay\%d in the
+ * element of @media, and create #GstRTSPStreams for them.
+ *
+ * Collect all dynamic elements, named dynpay\%d, and add them to
+ * the list of dynamic elements.
+ */
+void
+gst_rtsp_media_collect_streams (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstElement *element, *elem;
+ GstPad *pad;
+ gint i;
+ gboolean have_elem;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+ element = priv->element;
+
+ have_elem = TRUE;
+ for (i = 0; have_elem; i++) {
+ gchar *name;
+
+ have_elem = FALSE;
+
+ name = g_strdup_printf ("pay%d", i);
+ if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+ GST_INFO ("found stream %d with payloader %p", i, elem);
+
+ /* take the pad of the payloader */
+ pad = gst_element_get_static_pad (elem, "src");
+ /* create the stream */
+ gst_rtsp_media_create_stream (media, elem, pad);
+ gst_object_unref (pad);
+ gst_object_unref (elem);
+
+ have_elem = TRUE;
+ }
+ g_free (name);
+
+ name = g_strdup_printf ("dynpay%d", i);
+ if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+ /* a stream that will dynamically create pads to provide RTP packets */
+ GST_INFO ("found dynamic element %d, %p", i, elem);
+
+ g_mutex_lock (&priv->lock);
+ priv->dynamic = g_list_prepend (priv->dynamic, elem);
+ g_mutex_unlock (&priv->lock);
+
+ have_elem = TRUE;
+ }
+ g_free (name);
+ }
+}
+
+/**
+ * gst_rtsp_media_create_stream:
+ * @media: a #GstRTSPMedia
+ * @payloader: a #GstElement
+ * @srcpad: a source #GstPad
+ *
+ * Create a new stream in @media that provides RTP data on @srcpad.
+ * @srcpad should be a pad of an element inside @media->element.
+ *
+ * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
+ * as @media exists.
+ */
+GstRTSPStream *
+gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
+ GstPad * pad)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPStream *stream;
+ GstPad *srcpad;
+ gchar *name;
+ gint idx;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
+ g_return_val_if_fail (GST_IS_PAD (pad), NULL);
+ g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ idx = priv->streams->len;
+
+ GST_DEBUG ("media %p: creating stream with index %d", media, idx);
+
+ name = g_strdup_printf ("src_%u", idx);
+ srcpad = gst_ghost_pad_new (name, pad);
+ gst_pad_set_active (srcpad, TRUE);
+ gst_element_add_pad (priv->element, srcpad);
+ g_free (name);
+
+ stream = gst_rtsp_stream_new (idx, payloader, srcpad);
+ if (priv->pool)
+ gst_rtsp_stream_set_address_pool (stream, priv->pool);
+ gst_rtsp_stream_set_profiles (stream, priv->profiles);
+ gst_rtsp_stream_set_protocols (stream, priv->protocols);
+
+ g_ptr_array_add (priv->streams, stream);
+ g_mutex_unlock (&priv->lock);
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
+ NULL);
+
+ return stream;
+}
+
+static void
+gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
+{
+ GstRTSPMediaPrivate *priv;
+ GstPad *srcpad;
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* remove the ghostpad */
+ srcpad = gst_rtsp_stream_get_srcpad (stream);
+ gst_element_remove_pad (priv->element, srcpad);
+ gst_object_unref (srcpad);
+ /* now remove the stream */
+ g_object_ref (stream);
+ g_ptr_array_remove (priv->streams, stream);
+ g_mutex_unlock (&priv->lock);
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
+ stream, NULL);
+
+ g_object_unref (stream);
+}
+
+/**
+ * gst_rtsp_media_n_streams:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the number of streams in this media.
+ *
+ * Returns: The number of streams.
+ */
+guint
+gst_rtsp_media_n_streams (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->streams->len;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_get_stream:
+ * @media: a #GstRTSPMedia
+ * @idx: the stream index
+ *
+ * Retrieve the stream with index @idx from @media.
+ *
+ * Returns: (nullable) (transfer none): the #GstRTSPStream at index
+ * @idx or %NULL when a stream with that index did not exist.
+ */
+GstRTSPStream *
+gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPStream *res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (idx < priv->streams->len)
+ res = g_ptr_array_index (priv->streams, idx);
+ else
+ res = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_find_stream:
+ * @media: a #GstRTSPMedia
+ * @control: the control of the stream
+ *
+ * Find a stream in @media with @control as the control uri.
+ *
+ * Returns: (nullable) (transfer none): the #GstRTSPStream with
+ * control uri @control or %NULL when a stream with that control did
+ * not exist.
+ */
+GstRTSPStream *
+gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPStream *res;
+ gint i;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ g_return_val_if_fail (control != NULL, NULL);
+
+ priv = media->priv;
+
+ res = NULL;
+
+ g_mutex_lock (&priv->lock);
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *test;
+
+ test = g_ptr_array_index (priv->streams, i);
+ if (gst_rtsp_stream_has_control (test, control)) {
+ res = test;
+ break;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/* called with state-lock */
+static gboolean
+default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
+ GstRTSPRangeUnit unit)
+{
+ return gst_rtsp_range_convert_units (range, unit);
+}
+
+/**
+ * gst_rtsp_media_get_range_string:
+ * @media: a #GstRTSPMedia
+ * @play: for the PLAY request
+ * @unit: the unit to use for the string
+ *
+ * Get the current range as a string. @media must be prepared with
+ * gst_rtsp_media_prepare ().
+ *
+ * Returns: (transfer full): The range as a string, g_free() after usage.
+ */
+gchar *
+gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
+ GstRTSPRangeUnit unit)
+{
+ GstRTSPMediaClass *klass;
+ GstRTSPMediaPrivate *priv;
+ gchar *result;
+ GstRTSPTimeRange range;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ g_return_val_if_fail (klass->convert_range != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+ priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto not_prepared;
+
+ g_mutex_lock (&priv->lock);
+
+ /* Update the range value with current position/duration */
+ collect_media_stats (media);
+
+ /* make copy */
+ range = priv->range;
+
+ if (!play && priv->n_active > 0) {
+ range.min.type = GST_RTSP_TIME_NOW;
+ range.min.seconds = -1;
+ }
+ g_mutex_unlock (&priv->lock);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ if (!klass->convert_range (media, &range, unit))
+ goto conversion_failed;
+
+ result = gst_rtsp_range_to_string (&range);
+
+ return result;
+
+ /* ERRORS */
+not_prepared:
+ {
+ GST_WARNING ("media %p was not prepared", media);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return NULL;
+ }
+conversion_failed:
+ {
+ GST_WARNING ("range conversion to unit %d failed", unit);
+ return NULL;
+ }
+}
+
+static void
+stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
+{
+ gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
+}
+
+static void
+media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ GST_DEBUG ("media %p set blocked %d", media, blocked);
+ priv->blocked = blocked;
+ g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
+}
+
+static void
+gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->status = status;
+ GST_DEBUG ("setting new status to %d", status);
+ g_cond_broadcast (&priv->cond);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_status:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the status of @media. When @media is busy preparing, this function waits
+ * until @media is prepared or in error.
+ *
+ * Returns: the status of @media.
+ */
+GstRTSPMediaStatus
+gst_rtsp_media_get_status (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaStatus result;
+ gint64 end_time;
+
+ g_mutex_lock (&priv->lock);
+ end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
+ /* while we are preparing, wait */
+ while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
+ GST_DEBUG ("waiting for status change");
+ if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
+ GST_DEBUG ("timeout, assuming error status");
+ priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
+ }
+ }
+ /* could be success or error */
+ result = priv->status;
+ GST_DEBUG ("got status %d", result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_seek:
+ * @media: a #GstRTSPMedia
+ * @range: (transfer none): a #GstRTSPTimeRange
+ *
+ * Seek the pipeline of @media to @range. @media must be prepared with
+ * gst_rtsp_media_prepare().
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
+{
+ GstRTSPMediaClass *klass;
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+ GstClockTime start, stop;
+ GstSeekType start_type, stop_type;
+ GstQuery *query;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (range != NULL, FALSE);
+ g_return_val_if_fail (klass->convert_range != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto not_prepared;
+
+ /* Update the seekable state of the pipeline in case it changed */
+ query = gst_query_new_seeking (GST_FORMAT_TIME);
+ if (gst_element_query (priv->pipeline, query)) {
+ GstFormat format;
+ gboolean seekable;
+ gint64 start, end;
+
+ gst_query_parse_seeking (query, &format, &seekable, &start, &end);
+ priv->seekable = seekable;
+ }
+ gst_query_unref (query);
+
+ if (!priv->seekable)
+ goto not_seekable;
+
+ start_type = stop_type = GST_SEEK_TYPE_NONE;
+
+ if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
+ goto not_supported;
+ gst_rtsp_range_get_times (range, &start, &stop);
+
+ GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
+ GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
+
+ if (start != GST_CLOCK_TIME_NONE)
+ start_type = GST_SEEK_TYPE_SET;
+
+ if (priv->range_stop == stop)
+ stop = GST_CLOCK_TIME_NONE;
+ else if (stop != GST_CLOCK_TIME_NONE)
+ stop_type = GST_SEEK_TYPE_SET;
+
+ if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
+ GstSeekFlags flags;
+
+ GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+ if (priv->blocked)
+ media_streams_set_blocked (media, TRUE);
+
+ /* depends on the current playing state of the pipeline. We might need to
+ * queue this until we get EOS. */
+ flags = GST_SEEK_FLAG_FLUSH;
+
+ /* if range start was not supplied we must continue from current position.
+ * but since we're doing a flushing seek, let us query the current position
+ * so we end up at exactly the same position after the seek. */
+ if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
+ gint64 position;
+ gboolean ret = FALSE;
+
+ if (klass->query_position)
+ ret = klass->query_position (media, &position);
+
+ if (!ret) {
+ GST_WARNING ("position query failed");
+ } else {
+ GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (position));
+ start = position;
+ start_type = GST_SEEK_TYPE_SET;
+ flags |= GST_SEEK_FLAG_ACCURATE;
+ }
+ } else {
+ /* only set keyframe flag when modifying start */
+ if (start_type != GST_SEEK_TYPE_NONE)
+ flags |= GST_SEEK_FLAG_KEY_UNIT;
+ }
+
+ /* FIXME, we only do forwards playback, no trick modes yet */
+ res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
+ flags, start_type, start, stop_type, stop);
+
+ /* and block for the seek to complete */
+ GST_INFO ("done seeking %d", res);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ /* wait until pipeline is prerolled again, this will also collect stats */
+ if (!wait_preroll (media))
+ goto preroll_failed;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ GST_INFO ("prerolled again");
+ } else {
+ GST_INFO ("no seek needed");
+ res = TRUE;
+ }
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+
+ /* ERRORS */
+not_prepared:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("media %p is not prepared", media);
+ return FALSE;
+ }
+not_seekable:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("pipeline is not seekable");
+ return FALSE;
+ }
+not_supported:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("conversion to npt not supported");
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll after seek");
+ return FALSE;
+ }
+}
+
+static void
+stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
+{
+ *blocked &= gst_rtsp_stream_is_blocking (stream);
+}
+
+static gboolean
+media_streams_blocking (GstRTSPMedia * media)
+{
+ gboolean blocking = TRUE;
+
+ g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
+ &blocking);
+
+ return blocking;
+}
+
+static GstStateChangeReturn
+set_state (GstRTSPMedia * media, GstState state)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
+ media);
+ ret = gst_element_set_state (priv->pipeline, state);
+
+ return ret;
+}
+
+static GstStateChangeReturn
+set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ GST_INFO ("set target state to %s for media %p",
+ gst_element_state_get_name (state), media);
+ priv->target_state = state;
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
+ priv->target_state, NULL);
+
+ if (do_state)
+ ret = set_state (media, state);
+ else
+ ret = GST_STATE_CHANGE_SUCCESS;
+
+ return ret;
+}
+
+/* called with state-lock */
+static gboolean
+default_handle_message (GstRTSPMedia * media, GstMessage * message)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstMessageType type;
+
+ type = GST_MESSAGE_TYPE (message);
+
+ switch (type) {
+ case GST_MESSAGE_STATE_CHANGED:
+ break;
+ case GST_MESSAGE_BUFFERING:
+ {
+ gint percent;
+
+ gst_message_parse_buffering (message, &percent);
+
+ /* no state management needed for live pipelines */
+ if (priv->is_live)
+ break;
+
+ if (percent == 100) {
+ /* a 100% message means buffering is done */
+ priv->buffering = FALSE;
+ /* if the desired state is playing, go back */
+ if (priv->target_state == GST_STATE_PLAYING) {
+ GST_INFO ("Buffering done, setting pipeline to PLAYING");
+ set_state (media, GST_STATE_PLAYING);
+ } else {
+ GST_INFO ("Buffering done");
+ }
+ } else {
+ /* buffering busy */
+ if (priv->buffering == FALSE) {
+ if (priv->target_state == GST_STATE_PLAYING) {
+ /* we were not buffering but PLAYING, PAUSE the pipeline. */
+ GST_INFO ("Buffering, setting pipeline to PAUSED ...");
+ set_state (media, GST_STATE_PAUSED);
+ } else {
+ GST_INFO ("Buffering ...");
+ }
+ }
+ priv->buffering = TRUE;
+ }
+ break;
+ }
+ case GST_MESSAGE_LATENCY:
+ {
+ gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
+ break;
+ }
+ case GST_MESSAGE_ERROR:
+ {
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_error (message, &gerror, &debug);
+ GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
+ g_error_free (gerror);
+ g_free (debug);
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ break;
+ }
+ case GST_MESSAGE_WARNING:
+ {
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_warning (message, &gerror, &debug);
+ GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
+ g_error_free (gerror);
+ g_free (debug);
+ break;
+ }
+ case GST_MESSAGE_ELEMENT:
+ {
+ const GstStructure *s;
+
+ s = gst_message_get_structure (message);
+ if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
+ GST_DEBUG ("media received blocking message");
+ if (priv->blocked && media_streams_blocking (media)) {
+ GST_DEBUG ("media is blocking");
+ collect_media_stats (media);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ }
+ }
+ break;
+ }
+ case GST_MESSAGE_STREAM_STATUS:
+ break;
+ case GST_MESSAGE_ASYNC_DONE:
+ if (priv->adding) {
+ /* when we are dynamically adding pads, the addition of the udpsrc will
+ * temporarily produce ASYNC_DONE messages. We have to ignore them and
+ * wait for the final ASYNC_DONE after everything prerolled */
+ GST_INFO ("%p: ignoring ASYNC_DONE", media);
+ } else {
+ GST_INFO ("%p: got ASYNC_DONE", media);
+ collect_media_stats (media);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ }
+ break;
+ case GST_MESSAGE_EOS:
+ GST_INFO ("%p: got EOS", media);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
+ GST_DEBUG ("shutting down after EOS");
+ finish_unprepare (media);
+ }
+ break;
+ default:
+ GST_INFO ("%p: got message type %d (%s)", media, type,
+ gst_message_type_get_name (type));
+ break;
+ }
+ return TRUE;
+}
+
+static gboolean
+bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaClass *klass;
+ gboolean ret;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (klass->handle_message)
+ ret = klass->handle_message (media, message);
+ else
+ ret = FALSE;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return ret;
+}
+
+static void
+watch_destroyed (GstRTSPMedia * media)
+{
+ GST_DEBUG_OBJECT (media, "source destroyed");
+ g_object_unref (media);
+}
+
+static GstElement *
+find_payload_element (GstElement * payloader)
+{
+ GstElement *pay = NULL;
+
+ if (GST_IS_BIN (payloader)) {
+ GstIterator *iter;
+ GValue item = { 0 };
+
+ iter = gst_bin_iterate_recurse (GST_BIN (payloader));
+ while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
+ GstElement *element = (GstElement *) g_value_get_object (&item);
+ GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
+ const gchar *klass;
+
+ klass =
+ gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
+ if (klass == NULL)
+ continue;
+
+ if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
+ pay = gst_object_ref (element);
+ g_value_unset (&item);
+ break;
+ }
+ g_value_unset (&item);
+ }
+ gst_iterator_free (iter);
+ } else {
+ pay = g_object_ref (payloader);
+ }
+
+ return pay;
+}
+
+/* called from streaming threads */
+static void
+pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPStream *stream;
+ GstElement *pay;
+
+ /* find the real payload element */
+ pay = find_payload_element (element);
+ stream = gst_rtsp_media_create_stream (media, pay, pad);
+ gst_object_unref (pay);
+
+ GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
+ goto not_preparing;
+
+ g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
+
+ /* we will be adding elements below that will cause ASYNC_DONE to be
+ * posted in the bus. We want to ignore those messages until the
+ * pipeline really prerolled. */
+ priv->adding = TRUE;
+
+ /* join the element in the PAUSED state because this callback is
+ * called from the streaming thread and it is PAUSED */
+ if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+ priv->rtpbin, GST_STATE_PAUSED)) {
+ GST_WARNING ("failed to join bin element");
+ }
+
+ priv->adding = FALSE;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return;
+
+ /* ERRORS */
+not_preparing:
+ {
+ gst_rtsp_media_remove_stream (media, stream);
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("ignore pad because we are not preparing");
+ return;
+ }
+}
+
+static void
+pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPStream *stream;
+
+ stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
+ if (stream == NULL)
+ return;
+
+ GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
+
+ g_rec_mutex_lock (&priv->state_lock);
+ gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ gst_rtsp_media_remove_stream (media, stream);
+}
+
+static void
+remove_fakesink (GstRTSPMediaPrivate * priv)
+{
+ GstElement *fakesink;
+
+ g_mutex_lock (&priv->lock);
+ if ((fakesink = priv->fakesink))
+ gst_object_ref (fakesink);
+ priv->fakesink = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (fakesink) {
+ gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
+ gst_element_set_state (fakesink, GST_STATE_NULL);
+ gst_object_unref (fakesink);
+ GST_INFO ("removed fakesink");
+ }
+}
+
+static void
+no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ GST_INFO ("no more pads");
+ remove_fakesink (priv);
+}
+
+typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
+
+struct _DynPaySignalHandlers
+{
+ gulong pad_added_handler;
+ gulong pad_removed_handler;
+ gulong no_more_pads_handler;
+};
+
+static gboolean
+start_preroll (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ GST_INFO ("setting pipeline to PAUSED for media %p", media);
+ /* first go to PAUSED */
+ ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+
+ switch (ret) {
+ case GST_STATE_CHANGE_SUCCESS:
+ GST_INFO ("SUCCESS state change for media %p", media);
+ priv->seekable = TRUE;
+ break;
+ case GST_STATE_CHANGE_ASYNC:
+ GST_INFO ("ASYNC state change for media %p", media);
+ priv->seekable = TRUE;
+ break;
+ case GST_STATE_CHANGE_NO_PREROLL:
+ /* we need to go to PLAYING */
+ GST_INFO ("NO_PREROLL state change: live media %p", media);
+ /* FIXME we disable seeking for live streams for now. We should perform a
+ * seeking query in preroll instead */
+ priv->seekable = FALSE;
+ priv->is_live = TRUE;
+ /* start blocked to make sure nothing goes to the sink */
+ media_streams_set_blocked (media, TRUE);
+ ret = set_state (media, GST_STATE_PLAYING);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto state_failed;
+ break;
+ case GST_STATE_CHANGE_FAILURE:
+ goto state_failed;
+ }
+
+ return TRUE;
+
+state_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ return FALSE;
+ }
+}
+
+static gboolean
+wait_preroll (GstRTSPMedia * media)
+{
+ GstRTSPMediaStatus status;
+
+ GST_DEBUG ("wait to preroll pipeline");
+
+ /* wait until pipeline is prerolled */
+ status = gst_rtsp_media_get_status (media);
+ if (status == GST_RTSP_MEDIA_STATUS_ERROR)
+ goto preroll_failed;
+
+ return TRUE;
+
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ return FALSE;
+ }
+}
+
+static gboolean
+start_prepare (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ guint i;
+ GList *walk;
+
+ /* link streams we already have, other streams might appear when we have
+ * dynamic elements */
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream;
+
+ stream = g_ptr_array_index (priv->streams, i);
+
+ if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+ priv->rtpbin, GST_STATE_NULL)) {
+ goto join_bin_failed;
+ }
+ }
+
+ for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
+ GstElement *elem = walk->data;
+ DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
+
+ GST_INFO ("adding callbacks for dynamic element %p", elem);
+
+ handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
+ (GCallback) pad_added_cb, media);
+ handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
+ (GCallback) pad_removed_cb, media);
+ handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
+ (GCallback) no_more_pads_cb, media);
+
+ g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
+
+ /* we add a fakesink here in order to make the state change async. We remove
+ * the fakesink again in the no-more-pads callback. */
+ priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
+ gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
+ }
+
+ if (!start_preroll (media))
+ goto preroll_failed;
+
+ return FALSE;
+
+join_bin_failed:
+ {
+ GST_WARNING ("failed to join bin element");
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ return FALSE;
+ }
+}
+
+static gboolean
+default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+ GstBus *bus;
+ GMainContext *context;
+ GSource *source;
+
+ priv = media->priv;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ if (!klass->create_rtpbin)
+ goto no_create_rtpbin;
+
+ priv->rtpbin = klass->create_rtpbin (media);
+ if (priv->rtpbin != NULL) {
+ gboolean success = TRUE;
+
+ if (klass->setup_rtpbin)
+ success = klass->setup_rtpbin (media, priv->rtpbin);
+
+ if (success == FALSE) {
+ gst_object_unref (priv->rtpbin);
+ priv->rtpbin = NULL;
+ }
+ }
+ if (priv->rtpbin == NULL)
+ goto no_rtpbin;
+
+ priv->thread = thread;
+ context = (thread != NULL) ? (thread->context) : NULL;
+
+ bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
+
+ /* add the pipeline bus to our custom mainloop */
+ priv->source = gst_bus_create_watch (bus);
+ gst_object_unref (bus);
+
+ g_source_set_callback (priv->source, (GSourceFunc) bus_message,
+ g_object_ref (media), (GDestroyNotify) watch_destroyed);
+
+ priv->id = g_source_attach (priv->source, context);
+
+ /* add stuff to the bin */
+ gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
+
+ /* do remainder in context */
+ source = g_idle_source_new ();
+ g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
+ g_source_attach (source, context);
+ g_source_unref (source);
+
+ return TRUE;
+
+ /* ERRORS */
+no_create_rtpbin:
+ {
+ GST_ERROR ("no create_rtpbin function");
+ g_critical ("no create_rtpbin vmethod function set");
+ return FALSE;
+ }
+no_rtpbin:
+ {
+ GST_WARNING ("no rtpbin element");
+ g_warning ("failed to create element 'rtpbin', check your installation");
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_prepare:
+ * @media: a #GstRTSPMedia
+ * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
+ * bus handler or %NULL
+ *
+ * Prepare @media for streaming. This function will create the objects
+ * to manage the streaming. A pipeline must have been set on @media with
+ * gst_rtsp_media_take_pipeline().
+ *
+ * It will preroll the pipeline and collect vital information about the streams
+ * such as the duration.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ priv->prepare_count++;
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
+ priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto was_prepared;
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ goto is_preparing;
+
+ if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
+ goto not_unprepared;
+
+ if (!priv->reusable && priv->reused)
+ goto is_reused;
+
+ GST_INFO ("preparing media %p", media);
+
+ /* reset some variables */
+ priv->is_live = FALSE;
+ priv->seekable = FALSE;
+ priv->buffering = FALSE;
+
+ /* we're preparing now */
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->prepare) {
+ if (!klass->prepare (media, thread))
+ goto prepare_failed;
+ }
+
+wait_status:
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ /* now wait for all pads to be prerolled, FIXME, we should somehow be
+ * able to do this async so that we don't block the server thread. */
+ if (!wait_preroll (media))
+ goto preroll_failed;
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
+
+ GST_INFO ("object %p is prerolled", media);
+
+ return TRUE;
+
+ /* OK */
+is_preparing:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ goto wait_status;
+ }
+was_prepared:
+ {
+ GST_LOG ("media %p was prepared", media);
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return TRUE;
+ }
+ /* ERRORS */
+not_unprepared:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ GST_WARNING ("media %p was not unprepared", media);
+ priv->prepare_count--;
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+is_reused:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ priv->prepare_count--;
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("can not reuse media %p", media);
+ return FALSE;
+ }
+prepare_failed:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ priv->prepare_count--;
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_ERROR ("failed to prepare media");
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ gst_rtsp_media_unprepare (media);
+ return FALSE;
+ }
+}
+
+/* must be called with state-lock */
+static void
+finish_unprepare (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gint i;
+ GList *walk;
+
+ GST_DEBUG ("shutting down");
+
+ /* release the lock on shutdown, otherwise pad_added_cb might try to
+ * acquire the lock and then we deadlock */
+ g_rec_mutex_unlock (&priv->state_lock);
+ set_state (media, GST_STATE_NULL);
+ g_rec_mutex_lock (&priv->state_lock);
+ remove_fakesink (priv);
+
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream;
+
+ GST_INFO ("Removing elements of stream %d from pipeline", i);
+
+ stream = g_ptr_array_index (priv->streams, i);
+
+ gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
+ }
+
+ /* remove the pad signal handlers */
+ for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
+ GstElement *elem = walk->data;
+ DynPaySignalHandlers *handlers;
+
+ handlers =
+ g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
+ g_assert (handlers != NULL);
+
+ g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
+ g_signal_handler_disconnect (G_OBJECT (elem),
+ handlers->pad_removed_handler);
+ g_signal_handler_disconnect (G_OBJECT (elem),
+ handlers->no_more_pads_handler);
+
+ g_slice_free (DynPaySignalHandlers, handlers);
+ }
+
+ gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
+ priv->rtpbin = NULL;
+
+ if (priv->nettime)
+ gst_object_unref (priv->nettime);
+ priv->nettime = NULL;
+
+ priv->reused = TRUE;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
+
+ /* when the media is not reusable, this will effectively unref the media and
+ * recreate it */
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
+
+ /* the source has the last ref to the media */
+ if (priv->source) {
+ GST_DEBUG ("destroy source");
+ g_source_destroy (priv->source);
+ g_source_unref (priv->source);
+ }
+ if (priv->thread) {
+ GST_DEBUG ("stop thread");
+ gst_rtsp_thread_stop (priv->thread);
+ }
+}
+
+/* called with state-lock */
+static gboolean
+default_unprepare (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ if (priv->eos_shutdown) {
+ GST_DEBUG ("sending EOS for shutdown");
+ /* ref so that we don't disappear */
+ gst_element_send_event (priv->pipeline, gst_event_new_eos ());
+ /* we need to go to playing again for the EOS to propagate, normally in this
+ * state, nothing is receiving data from us anymore so this is ok. */
+ set_state (media, GST_STATE_PLAYING);
+ } else {
+ finish_unprepare (media);
+ }
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_media_unprepare:
+ * @media: a #GstRTSPMedia
+ *
+ * Unprepare @media. After this call, the media should be prepared again before
+ * it can be used again. If the media is set to be non-reusable, a new instance
+ * must be created.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_unprepare (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean success;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
+ goto was_unprepared;
+
+ priv->prepare_count--;
+ if (priv->prepare_count > 0)
+ goto is_busy;
+
+ GST_INFO ("unprepare media %p", media);
+ if (priv->blocked)
+ media_streams_set_blocked (media, FALSE);
+ set_target_state (media, GST_STATE_NULL, FALSE);
+ success = TRUE;
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
+ GstRTSPMediaClass *klass;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->unprepare)
+ success = klass->unprepare (media);
+ } else {
+ finish_unprepare (media);
+ }
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return success;
+
+was_unprepared:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("media %p was already unprepared", media);
+ return TRUE;
+ }
+is_busy:
+ {
+ GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return TRUE;
+ }
+}
+
+/* should be called with state-lock */
+static GstClock *
+get_clock_unlocked (GstRTSPMedia * media)
+{
+ if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
+ GST_DEBUG_OBJECT (media, "media was not prepared");
+ return NULL;
+ }
+ return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
+}
+
+/**
+ * gst_rtsp_media_get_clock:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the clock that is used by the pipeline in @media.
+ *
+ * @media must be prepared before this method returns a valid clock object.
+ *
+ * Returns: (transfer full): the #GstClock used by @media. unref after usage.
+ */
+GstClock *
+gst_rtsp_media_get_clock (GstRTSPMedia * media)
+{
+ GstClock *clock;
+ GstRTSPMediaPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ clock = get_clock_unlocked (media);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return clock;
+}
+
+/**
+ * gst_rtsp_media_get_base_time:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the base_time that is used by the pipeline in @media.
+ *
+ * @media must be prepared before this method returns a valid base_time.
+ *
+ * Returns: the base_time used by @media.
+ */
+GstClockTime
+gst_rtsp_media_get_base_time (GstRTSPMedia * media)
+{
+ GstClockTime result;
+ GstRTSPMediaPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto not_prepared;
+
+ result = gst_element_get_base_time (media->priv->pipeline);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return result;
+
+ /* ERRORS */
+not_prepared:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_DEBUG_OBJECT (media, "media was not prepared");
+ return GST_CLOCK_TIME_NONE;
+ }
+}
+
+/**
+ * gst_rtsp_media_get_time_provider:
+ * @media: a #GstRTSPMedia
+ * @address: (allow-none): an address or %NULL
+ * @port: a port or 0
+ *
+ * Get the #GstNetTimeProvider for the clock used by @media. The time provider
+ * will listen on @address and @port for client time requests.
+ *
+ * Returns: (transfer full): the #GstNetTimeProvider of @media.
+ */
+GstNetTimeProvider *
+gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
+ guint16 port)
+{
+ GstRTSPMediaPrivate *priv;
+ GstNetTimeProvider *provider = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->time_provider) {
+ if ((provider = priv->nettime) == NULL) {
+ GstClock *clock;
+
+ if (priv->time_provider && (clock = get_clock_unlocked (media))) {
+ provider = gst_net_time_provider_new (clock, address, port);
+ gst_object_unref (clock);
+
+ priv->nettime = provider;
+ }
+ }
+ }
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ if (provider)
+ gst_object_ref (provider);
+
+ return provider;
+}
+
+static gboolean
+default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
+{
+ return gst_rtsp_sdp_from_media (sdp, info, media);
+}
+
+/**
+ * gst_rtsp_media_setup_sdp:
+ * @media: a #GstRTSPMedia
+ * @sdp: (transfer none): a #GstSDPMessage
+ * @info: (transfer none): a #GstSDPInfo
+ *
+ * Add @media specific info to @sdp. @info is used to configure the connection
+ * information in the SDP.
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (sdp != NULL, FALSE);
+ g_return_val_if_fail (info != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ if (!klass->setup_sdp)
+ goto no_setup_sdp;
+
+ res = klass->setup_sdp (media, sdp, info);
+
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+
+ /* ERRORS */
+no_setup_sdp:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_ERROR ("no setup_sdp function");
+ g_critical ("no setup_sdp vmethod function set");
+ return FALSE;
+ }
+}
+
+static void
+do_set_seqnum (GstRTSPStream * stream)
+{
+ guint16 seq_num;
+ seq_num = gst_rtsp_stream_get_current_seqnum (stream);
+ gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
+}
+
+/* call with state_lock */
+gboolean
+default_suspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ switch (priv->suspend_mode) {
+ case GST_RTSP_SUSPEND_MODE_NONE:
+ GST_DEBUG ("media %p no suspend", media);
+ break;
+ case GST_RTSP_SUSPEND_MODE_PAUSE:
+ GST_DEBUG ("media %p suspend to PAUSED", media);
+ ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto state_failed;
+ break;
+ case GST_RTSP_SUSPEND_MODE_RESET:
+ GST_DEBUG ("media %p suspend to NULL", media);
+ ret = set_target_state (media, GST_STATE_NULL, TRUE);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto state_failed;
+ /* Because payloader needs to set the sequence number as
+ * monotonic, we need to preserve the sequence number
+ * after pause. (otherwise going from pause to play, which
+ * is actually from NULL to PLAY will create a new sequence
+ * number. */
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
+ break;
+ default:
+ break;
+ }
+
+ /* let the streams do the state changes freely, if any */
+ media_streams_set_blocked (media, FALSE);
+
+ return TRUE;
+
+ /* ERRORS */
+state_failed:
+ {
+ GST_WARNING ("failed changing pipeline's state for media %p", media);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_suspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Suspend @media. The state of the pipeline managed by @media is set to
+ * GST_STATE_NULL but all streams are kept. @media can be prepared again
+ * with gst_rtsp_media_unsuspend()
+ *
+ * @media must be prepared with gst_rtsp_media_prepare();
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_suspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ GST_FIXME ("suspend for dynamic pipelines needs fixing");
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto not_prepared;
+
+ /* don't attempt to suspend when something is busy */
+ if (priv->n_active > 0)
+ goto done;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->suspend) {
+ if (!klass->suspend (media))
+ goto suspend_failed;
+ }
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
+done:
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+not_prepared:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("media %p was not prepared", media);
+ return FALSE;
+ }
+suspend_failed:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ GST_WARNING ("failed to suspend media %p", media);
+ return FALSE;
+ }
+}
+
+/* call with state_lock */
+gboolean
+default_unsuspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ switch (priv->suspend_mode) {
+ case GST_RTSP_SUSPEND_MODE_NONE:
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ break;
+ case GST_RTSP_SUSPEND_MODE_PAUSE:
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ break;
+ case GST_RTSP_SUSPEND_MODE_RESET:
+ {
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+ if (!start_preroll (media))
+ goto start_failed;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ if (!wait_preroll (media))
+ goto preroll_failed;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ }
+ default:
+ break;
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+start_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_unsuspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Unsuspend @media if it was in a suspended state. This method does nothing
+ * when the media was not in the suspended state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_unsuspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto done;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->unsuspend) {
+ if (!klass->unsuspend (media))
+ goto unsuspend_failed;
+ }
+
+done:
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+unsuspend_failed:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("failed to unsuspend media %p", media);
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ return FALSE;
+ }
+}
+
+/* must be called with state-lock */
+static void
+media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ if (state == GST_STATE_NULL) {
+ gst_rtsp_media_unprepare (media);
+ } else {
+ GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
+ set_target_state (media, state, FALSE);
+ /* when we are buffering, don't update the state yet, this will be done
+ * when buffering finishes */
+ if (priv->buffering) {
+ GST_INFO ("Buffering busy, delay state change");
+ } else {
+ if (state == GST_STATE_PLAYING)
+ /* make sure pads are not blocking anymore when going to PLAYING */
+ media_streams_set_blocked (media, FALSE);
+
+ set_state (media, state);
+
+ /* and suspend after pause */
+ if (state == GST_STATE_PAUSED)
+ gst_rtsp_media_suspend (media);
+ }
+ }
+}
+
+/**
+ * gst_rtsp_media_set_pipeline_state:
+ * @media: a #GstRTSPMedia
+ * @state: the target state of the pipeline
+ *
+ * Set the state of the pipeline managed by @media to @state
+ */
+void
+gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
+{
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ g_rec_mutex_lock (&media->priv->state_lock);
+ media_set_pipeline_state_locked (media, state);
+ g_rec_mutex_unlock (&media->priv->state_lock);
+}
+
+/**
+ * gst_rtsp_media_set_state:
+ * @media: a #GstRTSPMedia
+ * @state: the target state of the media
+ * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
+ * a #GPtrArray of #GstRTSPStreamTransport pointers
+ *
+ * Set the state of @media to @state and for the transports in @transports.
+ *
+ * @media must be prepared with gst_rtsp_media_prepare();
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
+ GPtrArray * transports)
+{
+ GstRTSPMediaPrivate *priv;
+ gint i;
+ gboolean activate, deactivate, do_state;
+ gint old_active;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (transports != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
+ goto error_status;
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+ priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto not_prepared;
+
+ /* NULL and READY are the same */
+ if (state == GST_STATE_READY)
+ state = GST_STATE_NULL;
+
+ activate = deactivate = FALSE;
+
+ GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
+ media);
+
+ switch (state) {
+ case GST_STATE_NULL:
+ case GST_STATE_PAUSED:
+ /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
+ if (priv->target_state == GST_STATE_PLAYING)
+ deactivate = TRUE;
+ break;
+ case GST_STATE_PLAYING:
+ /* we're going to PLAYING, activate */
+ activate = TRUE;
+ break;
+ default:
+ break;
+ }
+ old_active = priv->n_active;
+
+ for (i = 0; i < transports->len; i++) {
+ GstRTSPStreamTransport *trans;
+
+ /* we need a non-NULL entry in the array */
+ trans = g_ptr_array_index (transports, i);
+ if (trans == NULL)
+ continue;
+
+ if (activate) {
+ if (gst_rtsp_stream_transport_set_active (trans, TRUE))
+ priv->n_active++;
+ } else if (deactivate) {
+ if (gst_rtsp_stream_transport_set_active (trans, FALSE))
+ priv->n_active--;
+ }
+ }
+
+ /* we just activated the first media, do the playing state change */
+ if (old_active == 0 && activate)
+ do_state = TRUE;
+ /* if we have no more active media, do the downward state changes */
+ else if (priv->n_active == 0)
+ do_state = TRUE;
+ else
+ do_state = FALSE;
+
+ GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
+ media, do_state);
+
+ if (priv->target_state != state) {
+ if (do_state)
+ media_set_pipeline_state_locked (media, state);
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
+ NULL);
+ }
+
+ /* remember where we are */
+ if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
+ old_active != priv->n_active))
+ collect_media_stats (media);
+
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+not_prepared:
+ {
+ GST_WARNING ("media %p was not prepared", media);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+error_status:
+ {
+ GST_WARNING ("media %p in error status while changing to state %d",
+ media, state);
+ if (state == GST_STATE_NULL) {
+ for (i = 0; i < transports->len; i++) {
+ GstRTSPStreamTransport *trans;
+
+ /* we need a non-NULL entry in the array */
+ trans = g_ptr_array_index (transports, i);
+ if (trans == NULL)
+ continue;
+
+ gst_rtsp_stream_transport_set_active (trans, FALSE);
+ }
+ priv->n_active = 0;
+ }
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtsprange.h>
+#include <gst/rtsp/gstrtspurl.h>
+#include <gst/net/gstnet.h>
+
+#ifndef __GST_RTSP_MEDIA_H__
+#define __GST_RTSP_MEDIA_H__
+
+G_BEGIN_DECLS
+
+/* types for the media */
+#define GST_TYPE_RTSP_MEDIA (gst_rtsp_media_get_type ())
+#define GST_IS_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA))
+#define GST_IS_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA))
+#define GST_RTSP_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
+#define GST_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia))
+#define GST_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
+#define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj))
+#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
+
+typedef struct _GstRTSPMedia GstRTSPMedia;
+typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
+typedef struct _GstRTSPMediaPrivate GstRTSPMediaPrivate;
+
+/**
+ * GstRTSPMediaStatus:
+ * @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
+ * @GST_RTSP_MEDIA_STATUS_UNPREPARING: media pipeline is busy doing a clean
+ * shutdown.
+ * @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
+ * @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
+ * @GST_RTSP_MEDIA_STATUS_SUSPENDED: media is suspended
+ * @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
+ *
+ * The state of the media pipeline.
+ */
+typedef enum {
+ GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
+ GST_RTSP_MEDIA_STATUS_UNPREPARING = 1,
+ GST_RTSP_MEDIA_STATUS_PREPARING = 2,
+ GST_RTSP_MEDIA_STATUS_PREPARED = 3,
+ GST_RTSP_MEDIA_STATUS_SUSPENDED = 4,
+ GST_RTSP_MEDIA_STATUS_ERROR = 5
+} GstRTSPMediaStatus;
+
+/**
+ * GstRTSPSuspendMode:
+ * @GST_RTSP_SUSPEND_MODE_NONE: Media is not suspended
+ * @GST_RTSP_SUSPEND_MODE_PAUSE: Media is PAUSED in suspend
+ * @GST_RTSP_SUSPEND_MODE_RESET: The media is set to NULL when suspended
+ *
+ * The suspend mode of the media pipeline. A media pipeline is suspended right
+ * after creating the SDP and when the client performs a PAUSED request.
+ */
+typedef enum {
+ GST_RTSP_SUSPEND_MODE_NONE = 0,
+ GST_RTSP_SUSPEND_MODE_PAUSE = 1,
+ GST_RTSP_SUSPEND_MODE_RESET = 2
+} GstRTSPSuspendMode;
+
+#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
+GType gst_rtsp_suspend_mode_get_type (void);
+
+#include "rtsp-stream.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-permissions.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-sdp.h"
+
+/**
+ * GstRTSPMedia:
+ *
+ * A class that contains the GStreamer element along with a list of
+ * #GstRTSPStream objects that can produce data.
+ *
+ * This object is usually created from a #GstRTSPMediaFactory.
+ */
+struct _GstRTSPMedia {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPMediaPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaClass:
+ * @handle_message: handle a message
+ * @prepare: the default implementation adds all elements and sets the
+ * pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING
+ * in case of NO_PREROLL elements).
+ * @unprepare: the default implementation sets the pipeline's state
+ * to GST_STATE_NULL and removes all elements.
+ * @suspend: the default implementation sets the pipeline's state to
+ * GST_STATE_NULL GST_STATE_PAUSED depending on the selected
+ * suspend mode.
+ * @unsuspend: the default implementation reverts the suspend operation.
+ * The pipeline will be prerolled again if it's state was
+ * set to GST_STATE_NULL in suspend.
+ * @convert_range: convert a range to the given unit
+ * @query_position: query the current position in the pipeline
+ * @query_stop: query when playback will stop
+ *
+ * The RTSP media class
+ */
+struct _GstRTSPMediaClass {
+ GObjectClass parent_class;
+
+ /* vmethods */
+ gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
+ gboolean (*prepare) (GstRTSPMedia *media, GstRTSPThread *thread);
+ gboolean (*unprepare) (GstRTSPMedia *media);
+ gboolean (*suspend) (GstRTSPMedia *media);
+ gboolean (*unsuspend) (GstRTSPMedia *media);
+ gboolean (*convert_range) (GstRTSPMedia *media, GstRTSPTimeRange *range,
+ GstRTSPRangeUnit unit);
+ gboolean (*query_position) (GstRTSPMedia *media, gint64 *position);
+ gboolean (*query_stop) (GstRTSPMedia *media, gint64 *stop);
+ GstElement * (*create_rtpbin) (GstRTSPMedia *media);
+ gboolean (*setup_rtpbin) (GstRTSPMedia *media, GstElement *rtpbin);
+ gboolean (*setup_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);
+
+ /* signals */
+ void (*new_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
+ void (*removed_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
+
+ void (*prepared) (GstRTSPMedia *media);
+ void (*unprepared) (GstRTSPMedia *media);
+
+ void (*target_state) (GstRTSPMedia *media, GstState state);
+ void (*new_state) (GstRTSPMedia *media, GstState state);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType gst_rtsp_media_get_type (void);
+
+/* creating the media */
+GstRTSPMedia * gst_rtsp_media_new (GstElement *element);
+GstElement * gst_rtsp_media_get_element (GstRTSPMedia *media);
+
+void gst_rtsp_media_take_pipeline (GstRTSPMedia *media, GstPipeline *pipeline);
+
+GstRTSPMediaStatus gst_rtsp_media_get_status (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_permissions (GstRTSPMedia *media,
+ GstRTSPPermissions *permissions);
+GstRTSPPermissions * gst_rtsp_media_get_permissions (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
+gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
+gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_profiles (GstRTSPMedia *media, GstRTSPProfile profiles);
+GstRTSPProfile gst_rtsp_media_get_profiles (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
+GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
+gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool);
+GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size);
+guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);
+
+void gst_rtsp_media_use_time_provider (GstRTSPMedia *media, gboolean time_provider);
+gboolean gst_rtsp_media_is_time_provider (GstRTSPMedia *media);
+GstNetTimeProvider * gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
+ const gchar *address, guint16 port);
+
+/* prepare the media for playback */
+gboolean gst_rtsp_media_prepare (GstRTSPMedia *media, GstRTSPThread *thread);
+gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media, GstRTSPSuspendMode mode);
+GstRTSPSuspendMode gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);
+
+gboolean gst_rtsp_media_suspend (GstRTSPMedia *media);
+gboolean gst_rtsp_media_unsuspend (GstRTSPMedia *media);
+
+gboolean gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info);
+
+/* creating streams */
+void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
+GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media,
+ GstElement *payloader,
+ GstPad *srcpad);
+
+/* dealing with the media */
+GstClock * gst_rtsp_media_get_clock (GstRTSPMedia *media);
+GstClockTime gst_rtsp_media_get_base_time (GstRTSPMedia *media);
+
+guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
+GstRTSPStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
+GstRTSPStream * gst_rtsp_media_find_stream (GstRTSPMedia *media, const gchar * control);
+
+gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
+gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media,
+ gboolean play,
+ GstRTSPRangeUnit unit);
+
+gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state,
+ GPtrArray *transports);
+void gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media,
+ GstState state);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MEDIA_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-mount-points
+ * @short_description: Map a path to media
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPClient
+ *
+ * A #GstRTSPMountPoints object maintains a relation between paths
+ * and #GstRTSPMediaFactory objects. This object is usually given to
+ * #GstRTSPClient and used to find the media attached to a path.
+ *
+ * With gst_rtsp_mount_points_add_factory () and
+ * gst_rtsp_mount_points_remove_factory(), factories can be added and
+ * removed.
+ *
+ * With gst_rtsp_mount_points_match() you can find the #GstRTSPMediaFactory
+ * object that completely matches the given path.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#include <string.h>
+
+#include "rtsp-mount-points.h"
+
+#define GST_RTSP_MOUNT_POINTS_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPointsPrivate))
+
+typedef struct
+{
+ gchar *path;
+ gint len;
+ GstRTSPMediaFactory *factory;
+} DataItem;
+
+static DataItem *
+data_item_new (gchar * path, gint len, GstRTSPMediaFactory * factory)
+{
+ DataItem *item;
+
+ item = g_slice_alloc (sizeof (DataItem));
+ item->path = path;
+ item->len = len;
+ item->factory = factory;
+
+ return item;
+}
+
+static void
+data_item_free (gpointer data)
+{
+ DataItem *item = data;
+
+ g_free (item->path);
+ g_object_unref (item->factory);
+ g_slice_free1 (sizeof (DataItem), item);
+}
+
+static void
+data_item_dump (gconstpointer a, gconstpointer prefix)
+{
+ const DataItem *item = a;
+
+ GST_DEBUG ("%s%s %p", (gchar *) prefix, item->path, item->factory);
+}
+
+static gint
+data_item_compare (gconstpointer a, gconstpointer b, gpointer user_data)
+{
+ const DataItem *item1 = a, *item2 = b;
+ gint res;
+
+ res = g_strcmp0 (item1->path, item2->path);
+
+ return res;
+}
+
+struct _GstRTSPMountPointsPrivate
+{
+ GMutex lock;
+ GSequence *mounts; /* protected by lock */
+ gboolean dirty;
+};
+
+G_DEFINE_TYPE (GstRTSPMountPoints, gst_rtsp_mount_points, G_TYPE_OBJECT);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
+#define GST_CAT_DEFAULT rtsp_media_debug
+
+static gchar *default_make_path (GstRTSPMountPoints * mounts,
+ const GstRTSPUrl * url);
+static void gst_rtsp_mount_points_finalize (GObject * obj);
+
+static void
+gst_rtsp_mount_points_class_init (GstRTSPMountPointsClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPMountPointsPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->finalize = gst_rtsp_mount_points_finalize;
+
+ klass->make_path = default_make_path;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmountpoints", 0,
+ "GstRTSPMountPoints");
+}
+
+static void
+gst_rtsp_mount_points_init (GstRTSPMountPoints * mounts)
+{
+ GstRTSPMountPointsPrivate *priv = GST_RTSP_MOUNT_POINTS_GET_PRIVATE (mounts);
+
+ GST_DEBUG_OBJECT (mounts, "created");
+
+ mounts->priv = priv;
+
+ g_mutex_init (&priv->lock);
+ priv->mounts = g_sequence_new (data_item_free);
+ priv->dirty = FALSE;
+}
+
+static void
+gst_rtsp_mount_points_finalize (GObject * obj)
+{
+ GstRTSPMountPoints *mounts = GST_RTSP_MOUNT_POINTS (obj);
+ GstRTSPMountPointsPrivate *priv = mounts->priv;
+
+ GST_DEBUG_OBJECT (mounts, "finalized");
+
+ g_sequence_free (priv->mounts);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_mount_points_parent_class)->finalize (obj);
+}
+
+/**
+ * gst_rtsp_mount_points_new:
+ *
+ * Make a new mount points object.
+ *
+ * Returns: (transfer full): a new #GstRTSPMountPoints
+ */
+GstRTSPMountPoints *
+gst_rtsp_mount_points_new (void)
+{
+ GstRTSPMountPoints *result;
+
+ result = g_object_new (GST_TYPE_RTSP_MOUNT_POINTS, NULL);
+
+ return result;
+}
+
+static gchar *
+default_make_path (GstRTSPMountPoints * mounts, const GstRTSPUrl * url)
+{
+ return g_strdup (url->abspath);
+}
+
+/**
+ * gst_rtsp_mount_points_make_path:
+ * @mounts: a #GstRTSPMountPoints
+ * @url: a #GstRTSPUrl
+ *
+ * Make a path string from @url.
+ *
+ * Returns: (transfer full): a path string for @url, g_free() after usage.
+ */
+gchar *
+gst_rtsp_mount_points_make_path (GstRTSPMountPoints * mounts,
+ const GstRTSPUrl * url)
+{
+ GstRTSPMountPointsClass *klass;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts), NULL);
+ g_return_val_if_fail (url != NULL, NULL);
+
+ klass = GST_RTSP_MOUNT_POINTS_GET_CLASS (mounts);
+
+ if (klass->make_path)
+ result = klass->make_path (mounts, url);
+ else
+ result = NULL;
+
+ return result;
+}
+
+static gboolean
+has_prefix (DataItem * str, DataItem * prefix)
+{
+ /* prefix needs to be smaller than str */
+ if (str->len < prefix->len)
+ return FALSE;
+
+ /* if str is larger, it there should be a / following the prefix */
+ if (str->len > prefix->len && str->path[prefix->len] != '/')
+ return FALSE;
+
+ return strncmp (str->path, prefix->path, prefix->len) == 0;
+}
+
+/**
+ * gst_rtsp_mount_points_match:
+ * @mounts: a #GstRTSPMountPoints
+ * @path: a mount point
+ * @matched: (out) (allow-none): the amount of @path matched
+ *
+ * Find the factory in @mounts that has the longest match with @path.
+ *
+ * If @matched is %NULL, @path will match the factory exactly otherwise
+ * the amount of characters that matched is returned in @matched.
+ *
+ * Returns: (transfer full): the #GstRTSPMediaFactory for @path.
+ * g_object_unref() after usage.
+ */
+GstRTSPMediaFactory *
+gst_rtsp_mount_points_match (GstRTSPMountPoints * mounts,
+ const gchar * path, gint * matched)
+{
+ GstRTSPMountPointsPrivate *priv;
+ GstRTSPMediaFactory *result = NULL;
+ GSequenceIter *iter, *best;
+ DataItem item, *ritem;
+
+ g_return_val_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts), NULL);
+ g_return_val_if_fail (path != NULL, NULL);
+
+ priv = mounts->priv;
+
+ item.path = (gchar *) path;
+ item.len = strlen (path);
+
+ g_mutex_lock (&priv->lock);
+ if (priv->dirty) {
+ g_sequence_sort (priv->mounts, data_item_compare, mounts);
+ g_sequence_foreach (priv->mounts, (GFunc) data_item_dump,
+ (gpointer) "sort :");
+ priv->dirty = FALSE;
+ }
+
+ /* find the location of the media in the hashtable we only use the absolute
+ * path of the uri to find a media factory. If the factory depends on other
+ * properties found in the url, this method should be overridden. */
+ iter = g_sequence_get_begin_iter (priv->mounts);
+ best = NULL;
+ while (!g_sequence_iter_is_end (iter)) {
+ ritem = g_sequence_get (iter);
+
+ data_item_dump (ritem, "inspect: ");
+
+ if (best == NULL) {
+ if (has_prefix (&item, ritem)) {
+ data_item_dump (ritem, "prefix: ");
+ best = iter;
+ }
+ } else {
+ if (!has_prefix (&item, ritem))
+ break;
+
+ best = iter;
+ data_item_dump (ritem, "new best: ");
+ }
+ iter = g_sequence_iter_next (iter);
+ }
+ if (best) {
+ ritem = g_sequence_get (best);
+ data_item_dump (ritem, "result: ");
+ if (matched || ritem->len == item.len) {
+ result = g_object_ref (ritem->factory);
+ if (matched)
+ *matched = ritem->len;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ GST_INFO ("found media factory %p for path %s", result, path);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_mount_points_add_factory:
+ * @mounts: a #GstRTSPMountPoints
+ * @path: a mount point
+ * @factory: (transfer full): a #GstRTSPMediaFactory
+ *
+ * Attach @factory to the mount point @path in @mounts.
+ *
+ * @path is of the form (/node)+. Any previous mount point will be freed.
+ *
+ * Ownership is taken of the reference on @factory so that @factory should not be
+ * used after calling this function.
+ */
+void
+gst_rtsp_mount_points_add_factory (GstRTSPMountPoints * mounts,
+ const gchar * path, GstRTSPMediaFactory * factory)
+{
+ GstRTSPMountPointsPrivate *priv;
+ DataItem *item;
+
+ g_return_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts));
+ g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+ g_return_if_fail (path != NULL);
+
+ priv = mounts->priv;
+
+ item = data_item_new (g_strdup (path), strlen (path), factory);
+
+ GST_INFO ("adding media factory %p for path %s", factory, path);
+
+ g_mutex_lock (&priv->lock);
+ g_sequence_append (priv->mounts, item);
+ priv->dirty = TRUE;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_mount_points_remove_factory:
+ * @mounts: a #GstRTSPMountPoints
+ * @path: a mount point
+ *
+ * Remove the #GstRTSPMediaFactory associated with @path in @mounts.
+ */
+void
+gst_rtsp_mount_points_remove_factory (GstRTSPMountPoints * mounts,
+ const gchar * path)
+{
+ GstRTSPMountPointsPrivate *priv;
+ DataItem item;
+ GSequenceIter *iter;
+
+ g_return_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts));
+ g_return_if_fail (path != NULL);
+
+ priv = mounts->priv;
+
+ item.path = (gchar *) path;
+
+ GST_INFO ("removing media factory for path %s", path);
+
+ g_mutex_lock (&priv->lock);
+ if (priv->dirty) {
+ g_sequence_sort (priv->mounts, data_item_compare, mounts);
+ priv->dirty = FALSE;
+ }
+ iter = g_sequence_lookup (priv->mounts, &item, data_item_compare, mounts);
+ if (iter) {
+ g_sequence_remove (iter);
+ priv->dirty = TRUE;
+ }
+ g_mutex_unlock (&priv->lock);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include "rtsp-media-factory.h"
+
+#ifndef __GST_RTSP_MOUNT_POINTS_H__
+#define __GST_RTSP_MOUNT_POINTS_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_MOUNT_POINTS (gst_rtsp_mount_points_get_type ())
+#define GST_IS_RTSP_MOUNT_POINTS(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MOUNT_POINTS))
+#define GST_IS_RTSP_MOUNT_POINTS_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MOUNT_POINTS))
+#define GST_RTSP_MOUNT_POINTS_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPointsClass))
+#define GST_RTSP_MOUNT_POINTS(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPoints))
+#define GST_RTSP_MOUNT_POINTS_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPointsClass))
+#define GST_RTSP_MOUNT_POINTS_CAST(obj) ((GstRTSPMountPoints*)(obj))
+#define GST_RTSP_MOUNT_POINTS_CLASS_CAST(klass) ((GstRTSPMountPointsClass*)(klass))
+
+typedef struct _GstRTSPMountPoints GstRTSPMountPoints;
+typedef struct _GstRTSPMountPointsClass GstRTSPMountPointsClass;
+typedef struct _GstRTSPMountPointsPrivate GstRTSPMountPointsPrivate;
+
+/**
+ * GstRTSPMountPoints:
+ *
+ * Creates a #GstRTSPMediaFactory object for a given url.
+ */
+struct _GstRTSPMountPoints {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPMountPointsPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMountPointsClass:
+ * @make_path: make a path from the given url.
+ *
+ * The class for the media mounts object.
+ */
+struct _GstRTSPMountPointsClass {
+ GObjectClass parent_class;
+
+ gchar * (*make_path) (GstRTSPMountPoints *mounts,
+ const GstRTSPUrl *url);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_mount_points_get_type (void);
+
+/* creating a mount points */
+GstRTSPMountPoints * gst_rtsp_mount_points_new (void);
+
+gchar * gst_rtsp_mount_points_make_path (GstRTSPMountPoints *mounts,
+ const GstRTSPUrl * url);
+/* finding a media factory */
+GstRTSPMediaFactory * gst_rtsp_mount_points_match (GstRTSPMountPoints *mounts,
+ const gchar *path,
+ gint * matched);
+/* managing media to a mount point */
+void gst_rtsp_mount_points_add_factory (GstRTSPMountPoints *mounts,
+ const gchar *path,
+ GstRTSPMediaFactory *factory);
+void gst_rtsp_mount_points_remove_factory (GstRTSPMountPoints *mounts,
+ const gchar *path);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MOUNT_POINTS_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-params
+ * @short_description: Param get and set implementation
+ * @see_also: #GstRTSPClient
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include <string.h>
+
+#include "rtsp-params.h"
+
+/**
+ * gst_rtsp_params_set:
+ * @client: a #GstRTSPClient
+ * @ctx: (transfer none): a #GstRTSPContext
+ *
+ * Set parameters (not implemented yet)
+ *
+ * Returns: a #GstRTSPResult
+ */
+GstRTSPResult
+gst_rtsp_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPStatusCode code;
+
+ /* FIXME, actually parse the request based on the mime type and try to repond
+ * with a list of the parameters */
+ code = GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD;
+
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_params_get:
+ * @client: a #GstRTSPClient
+ * @ctx: (transfer none): a #GstRTSPContext
+ *
+ * Get parameters (not implemented yet)
+ *
+ * Returns: a #GstRTSPResult
+ */
+GstRTSPResult
+gst_rtsp_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPStatusCode code;
+
+ /* FIXME, actually parse the request based on the mime type and try to repond
+ * with a list of the parameters */
+ code = GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD;
+
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ return GST_RTSP_OK;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp/gstrtspurl.h>
+#include <gst/rtsp/gstrtspmessage.h>
+
+#ifndef __GST_RTSP_PARAMS_H__
+#define __GST_RTSP_PARAMS_H__
+
+#include "rtsp-client.h"
+#include "rtsp-session.h"
+
+G_BEGIN_DECLS
+
+GstRTSPResult gst_rtsp_params_set (GstRTSPClient * client, GstRTSPContext * ctx);
+GstRTSPResult gst_rtsp_params_get (GstRTSPClient * client, GstRTSPContext * ctx);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_PARAMS_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-permissions
+ * @short_description: Roles and associated permissions
+ * @see_also: #GstRTSPToken, #GstRTSPAuth
+ *
+ * The #GstRTSPPermissions object contains an array of roles and associated
+ * permissions. The roles are represented with a string and the permissions with
+ * a generic #GstStructure.
+ *
+ * The permissions are deliberately kept generic. The possible values of the
+ * roles and #GstStructure keys and values are only determined by the #GstRTSPAuth
+ * object that performs the checks on the permissions and the current
+ * #GstRTSPToken.
+ *
+ * As a convenience function, gst_rtsp_permissions_is_allowed() can be used to
+ * check if the permissions contains a role that contains the boolean value
+ * %TRUE for the the given key.
+ *
+ * Last reviewed on 2013-07-15 (1.0.0)
+ */
+
+#include <string.h>
+
+#include "rtsp-permissions.h"
+
+typedef struct _GstRTSPPermissionsImpl
+{
+ GstRTSPPermissions permissions;
+
+ /* Roles, array of GstStructure */
+ GPtrArray *roles;
+} GstRTSPPermissionsImpl;
+
+static void
+free_structure (GstStructure * structure)
+{
+ gst_structure_set_parent_refcount (structure, NULL);
+ gst_structure_free (structure);
+}
+
+//GST_DEBUG_CATEGORY_STATIC (rtsp_permissions_debug);
+//#define GST_CAT_DEFAULT rtsp_permissions_debug
+
+GST_DEFINE_MINI_OBJECT_TYPE (GstRTSPPermissions, gst_rtsp_permissions);
+
+static void gst_rtsp_permissions_init (GstRTSPPermissionsImpl * permissions);
+
+static void
+_gst_rtsp_permissions_free (GstRTSPPermissions * permissions)
+{
+ GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+
+ g_ptr_array_free (impl->roles, TRUE);
+
+ g_slice_free1 (sizeof (GstRTSPPermissionsImpl), permissions);
+}
+
+static GstRTSPPermissions *
+_gst_rtsp_permissions_copy (GstRTSPPermissionsImpl * permissions)
+{
+ GstRTSPPermissionsImpl *copy;
+ guint i;
+
+ copy = (GstRTSPPermissionsImpl *) gst_rtsp_permissions_new ();
+
+ for (i = 0; i < permissions->roles->len; i++) {
+ GstStructure *entry = g_ptr_array_index (permissions->roles, i);
+ GstStructure *entry_copy = gst_structure_copy (entry);
+
+ gst_structure_set_parent_refcount (entry_copy,
+ ©->permissions.mini_object.refcount);
+ g_ptr_array_add (copy->roles, entry_copy);
+ }
+
+ return GST_RTSP_PERMISSIONS (copy);
+}
+
+static void
+gst_rtsp_permissions_init (GstRTSPPermissionsImpl * permissions)
+{
+ gst_mini_object_init (GST_MINI_OBJECT_CAST (permissions), 0,
+ GST_TYPE_RTSP_PERMISSIONS,
+ (GstMiniObjectCopyFunction) _gst_rtsp_permissions_copy, NULL,
+ (GstMiniObjectFreeFunction) _gst_rtsp_permissions_free);
+
+ permissions->roles =
+ g_ptr_array_new_with_free_func ((GDestroyNotify) free_structure);
+}
+
+/**
+ * gst_rtsp_permissions_new:
+ *
+ * Create a new empty Authorization permissions.
+ *
+ * Returns: (transfer full): a new empty authorization permissions.
+ */
+GstRTSPPermissions *
+gst_rtsp_permissions_new (void)
+{
+ GstRTSPPermissionsImpl *permissions;
+
+ permissions = g_slice_new0 (GstRTSPPermissionsImpl);
+ gst_rtsp_permissions_init (permissions);
+
+ return GST_RTSP_PERMISSIONS (permissions);
+}
+
+/**
+ * gst_rtsp_permissions_add_role:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ * @fieldname: the first field name
+ * @...: additional arguments
+ *
+ * Add a new @role to @permissions with the given variables. The fields
+ * are the same layout as gst_structure_new().
+ */
+void
+gst_rtsp_permissions_add_role (GstRTSPPermissions * permissions,
+ const gchar * role, const gchar * fieldname, ...)
+{
+ va_list var_args;
+
+ va_start (var_args, fieldname);
+ gst_rtsp_permissions_add_role_valist (permissions, role, fieldname, var_args);
+ va_end (var_args);
+}
+
+/**
+ * gst_rtsp_permissions_add_role_valist:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ * @fieldname: the first field name
+ * @var_args: additional fields to add
+ *
+ * Add a new @role to @permissions with the given variables. Structure fields
+ * are set according to the varargs in a manner similar to gst_structure_new().
+ */
+void
+gst_rtsp_permissions_add_role_valist (GstRTSPPermissions * permissions,
+ const gchar * role, const gchar * fieldname, va_list var_args)
+{
+ GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+ GstStructure *structure;
+ guint i, len;
+
+ g_return_if_fail (GST_IS_RTSP_PERMISSIONS (permissions));
+ g_return_if_fail (gst_mini_object_is_writable (&permissions->mini_object));
+ g_return_if_fail (role != NULL);
+ g_return_if_fail (fieldname != NULL);
+
+ structure = gst_structure_new_valist (role, fieldname, var_args);
+ g_return_if_fail (structure != NULL);
+
+ len = impl->roles->len;
+ for (i = 0; i < len; i++) {
+ GstStructure *entry = g_ptr_array_index (impl->roles, i);
+
+ if (gst_structure_has_name (entry, role)) {
+ g_ptr_array_remove_index_fast (impl->roles, i);
+ break;
+ }
+ }
+
+ gst_structure_set_parent_refcount (structure,
+ &impl->permissions.mini_object.refcount);
+ g_ptr_array_add (impl->roles, structure);
+}
+
+/**
+ * gst_rtsp_permissions_remove_role:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ *
+ * Remove all permissions for @role in @permissions.
+ */
+void
+gst_rtsp_permissions_remove_role (GstRTSPPermissions * permissions,
+ const gchar * role)
+{
+ GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+ guint i, len;
+
+ g_return_if_fail (GST_IS_RTSP_PERMISSIONS (permissions));
+ g_return_if_fail (gst_mini_object_is_writable (&permissions->mini_object));
+ g_return_if_fail (role != NULL);
+
+ len = impl->roles->len;
+ for (i = 0; i < len; i++) {
+ GstStructure *entry = g_ptr_array_index (impl->roles, i);
+
+ if (gst_structure_has_name (entry, role)) {
+ g_ptr_array_remove_index_fast (impl->roles, i);
+ break;
+ }
+ }
+}
+
+/**
+ * gst_rtsp_permissions_get_role:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ *
+ * Get all permissions for @role in @permissions.
+ *
+ * Returns: (transfer none): the structure with permissions for @role. It
+ * remains valid for as long as @permissions is valid.
+ */
+const GstStructure *
+gst_rtsp_permissions_get_role (GstRTSPPermissions * permissions,
+ const gchar * role)
+{
+ GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+ guint i, len;
+
+ g_return_val_if_fail (GST_IS_RTSP_PERMISSIONS (permissions), NULL);
+ g_return_val_if_fail (role != NULL, NULL);
+
+ len = impl->roles->len;
+ for (i = 0; i < len; i++) {
+ GstStructure *entry = g_ptr_array_index (impl->roles, i);
+
+ if (gst_structure_has_name (entry, role))
+ return entry;
+ }
+ return NULL;
+}
+
+/**
+ * gst_rtsp_permissions_is_allowed:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ * @permission: a permission
+ *
+ * Check if @role in @permissions is given permission for @permission.
+ *
+ * Returns: %TRUE if @role is allowed @permission.
+ */
+gboolean
+gst_rtsp_permissions_is_allowed (GstRTSPPermissions * permissions,
+ const gchar * role, const gchar * permission)
+{
+ const GstStructure *str;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_PERMISSIONS (permissions), FALSE);
+ g_return_val_if_fail (role != NULL, FALSE);
+ g_return_val_if_fail (permission != NULL, FALSE);
+
+ str = gst_rtsp_permissions_get_role (permissions, role);
+ if (str == NULL)
+ return FALSE;
+
+ if (!gst_structure_get_boolean (str, permission, &result))
+ result = FALSE;
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_PERMISSIONS_H__
+#define __GST_RTSP_PERMISSIONS_H__
+
+typedef struct _GstRTSPPermissions GstRTSPPermissions;
+
+G_BEGIN_DECLS
+
+GType gst_rtsp_permissions_get_type (void);
+
+#define GST_TYPE_RTSP_PERMISSIONS (gst_rtsp_permissions_get_type ())
+#define GST_IS_RTSP_PERMISSIONS(obj) (GST_IS_MINI_OBJECT_TYPE (obj, GST_TYPE_RTSP_PERMISSIONS))
+#define GST_RTSP_PERMISSIONS_CAST(obj) ((GstRTSPPermissions*)(obj))
+#define GST_RTSP_PERMISSIONS(obj) (GST_RTSP_PERMISSIONS_CAST(obj))
+
+/**
+ * GstRTSPPermissions:
+ *
+ * The opaque permissions structure. It is used to define the permissions
+ * of objects in different roles.
+ */
+struct _GstRTSPPermissions {
+ GstMiniObject mini_object;
+};
+
+/* refcounting */
+/**
+ * gst_rtsp_permissions_ref:
+ * @permissions: The permissions to refcount
+ *
+ * Increase the refcount of this permissions.
+ *
+ * Returns: (transfer full): @permissions (for convenience when doing assignments)
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC GstRTSPPermissions * gst_rtsp_permissions_ref (GstRTSPPermissions * permissions);
+#endif
+
+static inline GstRTSPPermissions *
+gst_rtsp_permissions_ref (GstRTSPPermissions * permissions)
+{
+ return (GstRTSPPermissions *) gst_mini_object_ref (GST_MINI_OBJECT_CAST (permissions));
+}
+
+/**
+ * gst_rtsp_permissions_unref:
+ * @permissions: (transfer full): the permissions to refcount
+ *
+ * Decrease the refcount of an permissions, freeing it if the refcount reaches 0.
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC void gst_rtsp_permissions_unref (GstRTSPPermissions * permissions);
+#endif
+
+static inline void
+gst_rtsp_permissions_unref (GstRTSPPermissions * permissions)
+{
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (permissions));
+}
+
+
+GstRTSPPermissions * gst_rtsp_permissions_new (void);
+
+void gst_rtsp_permissions_add_role (GstRTSPPermissions *permissions,
+ const gchar *role,
+ const gchar *fieldname, ...);
+void gst_rtsp_permissions_add_role_valist (GstRTSPPermissions *permissions,
+ const gchar *role,
+ const gchar *fieldname,
+ va_list var_args);
+void gst_rtsp_permissions_remove_role (GstRTSPPermissions *permissions,
+ const gchar *role);
+const GstStructure * gst_rtsp_permissions_get_role (GstRTSPPermissions *permissions,
+ const gchar *role);
+
+gboolean gst_rtsp_permissions_is_allowed (GstRTSPPermissions *permissions,
+ const gchar *role, const gchar *permission);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_PERMISSIONS_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-sdp
+ * @short_description: Make SDP messages
+ * @see_also: #GstRTSPMedia
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include <string.h>
+
+#include <gst/sdp/gstmikey.h>
+
+#include "rtsp-sdp.h"
+
+#define AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
+static gboolean
+get_info_from_tags (GstPad * pad, GstEvent ** event, gpointer user_data)
+{
+ GstSDPMedia *media = (GstSDPMedia *) user_data;
+
+ if (GST_EVENT_TYPE (*event) == GST_EVENT_TAG) {
+ GstTagList *tags;
+ guint bitrate = 0;
+
+ gst_event_parse_tag (*event, &tags);
+
+ if (gst_tag_list_get_scope (tags) != GST_TAG_SCOPE_STREAM)
+ return TRUE;
+
+ if (!gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE,
+ &bitrate) || bitrate == 0)
+ if (!gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &bitrate) ||
+ bitrate == 0)
+ return TRUE;
+
+ /* set bandwidth (kbits/s) */
+ gst_sdp_media_add_bandwidth (media, GST_SDP_BWTYPE_AS, bitrate / 1000);
+
+ return FALSE;
+
+ }
+
+ return TRUE;
+}
+
+static void
+update_sdp_from_tags (GstRTSPStream * stream, GstSDPMedia * stream_media)
+{
+ GstPad *src_pad;
+
+ src_pad = gst_rtsp_stream_get_srcpad (stream);
+
+ gst_pad_sticky_events_foreach (src_pad, get_info_from_tags, stream_media);
+
+ gst_object_unref (src_pad);
+}
+
+static guint8
+enc_key_length_from_cipher_name (const gchar * cipher)
+{
+ if (g_strcmp0 (cipher, "aes-128-icm") == 0)
+ return AES_128_KEY_LEN;
+ else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
+ return AES_256_KEY_LEN;
+ else {
+ GST_ERROR ("encryption algorithm '%s' not supported", cipher);
+ return 0;
+ }
+}
+
+static guint8
+auth_key_length_from_auth_name (const gchar * auth)
+{
+ if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
+ return HMAC_32_KEY_LEN;
+ else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
+ return HMAC_80_KEY_LEN;
+ else {
+ GST_ERROR ("authentication algorithm '%s' not supported", auth);
+ return 0;
+ }
+}
+
+static void
+make_media (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPMedia * media,
+ GstRTSPStream * stream, GstStructure * s, GstRTSPProfile profile)
+{
+ GstSDPMedia *smedia;
+ const gchar *caps_str, *caps_enc, *caps_params;
+ gchar *tmp;
+ gint caps_pt, caps_rate;
+ guint n_fields, j;
+ gboolean first;
+ GString *fmtp;
+ GstRTSPLowerTrans ltrans;
+ GSocketFamily family;
+ const gchar *addrtype, *proto;
+ gchar *address;
+ guint ttl;
+
+ gst_sdp_media_new (&smedia);
+
+ /* get media type and payload for the m= line */
+ caps_str = gst_structure_get_string (s, "media");
+ gst_sdp_media_set_media (smedia, caps_str);
+
+ gst_structure_get_int (s, "payload", &caps_pt);
+ tmp = g_strdup_printf ("%d", caps_pt);
+ gst_sdp_media_add_format (smedia, tmp);
+ g_free (tmp);
+
+ gst_sdp_media_set_port_info (smedia, 0, 1);
+
+ switch (profile) {
+ case GST_RTSP_PROFILE_AVP:
+ proto = "RTP/AVP";
+ break;
+ case GST_RTSP_PROFILE_AVPF:
+ proto = "RTP/AVPF";
+ break;
+ case GST_RTSP_PROFILE_SAVP:
+ proto = "RTP/SAVP";
+ break;
+ case GST_RTSP_PROFILE_SAVPF:
+ proto = "RTP/SAVPF";
+ break;
+ default:
+ proto = "udp";
+ break;
+ }
+ gst_sdp_media_set_proto (smedia, proto);
+
+ if (info->is_ipv6) {
+ addrtype = "IP6";
+ family = G_SOCKET_FAMILY_IPV6;
+ } else {
+ addrtype = "IP4";
+ family = G_SOCKET_FAMILY_IPV4;
+ }
+
+ ltrans = gst_rtsp_stream_get_protocols (stream);
+ if (ltrans == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ GstRTSPAddress *addr;
+
+ addr = gst_rtsp_stream_get_multicast_address (stream, family);
+ if (addr == NULL)
+ goto no_multicast;
+
+ address = g_strdup (addr->address);
+ ttl = addr->ttl;
+ gst_rtsp_address_free (addr);
+ } else {
+ ttl = 16;
+ if (info->is_ipv6)
+ address = g_strdup ("::");
+ else
+ address = g_strdup ("0.0.0.0");
+ }
+
+ /* for the c= line */
+ gst_sdp_media_add_connection (smedia, "IN", addrtype, address, ttl, 1);
+ g_free (address);
+
+ /* get clock-rate, media type and params for the rtpmap attribute */
+ gst_structure_get_int (s, "clock-rate", &caps_rate);
+ caps_enc = gst_structure_get_string (s, "encoding-name");
+ caps_params = gst_structure_get_string (s, "encoding-params");
+
+ if (caps_enc) {
+ if (caps_params)
+ tmp = g_strdup_printf ("%d %s/%d/%s", caps_pt, caps_enc, caps_rate,
+ caps_params);
+ else
+ tmp = g_strdup_printf ("%d %s/%d", caps_pt, caps_enc, caps_rate);
+
+ gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
+ g_free (tmp);
+ }
+
+ /* the config uri */
+ tmp = gst_rtsp_stream_get_control (stream);
+ gst_sdp_media_add_attribute (smedia, "control", tmp);
+ g_free (tmp);
+
+
+ /* check for srtp */
+ do {
+ GstBuffer *srtpkey;
+ const GValue *val;
+ const gchar *srtpcipher, *srtpauth, *srtcpcipher, *srtcpauth;
+ GstMIKEYMessage *msg;
+ GstMIKEYPayload *payload, *pkd;
+ GBytes *bytes;
+ GstMapInfo info;
+ const guint8 *data;
+ gsize size;
+ gchar *base64;
+ guint8 byte;
+ guint32 ssrc;
+
+ val = gst_structure_get_value (s, "srtp-key");
+ if (val == NULL)
+ break;
+
+ srtpkey = gst_value_get_buffer (val);
+ if (srtpkey == NULL)
+ break;
+
+ srtpcipher = gst_structure_get_string (s, "srtp-cipher");
+ srtpauth = gst_structure_get_string (s, "srtp-auth");
+ srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
+ srtcpauth = gst_structure_get_string (s, "srtcp-auth");
+
+ if (srtpcipher == NULL || srtpauth == NULL || srtcpcipher == NULL ||
+ srtcpauth == NULL)
+ break;
+
+ msg = gst_mikey_message_new ();
+ /* unencrypted MIKEY message, we send this over TLS so this is allowed */
+ gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
+ FALSE, GST_MIKEY_PRF_MIKEY_1, 0, GST_MIKEY_MAP_TYPE_SRTP);
+ /* add policy '0' for our SSRC */
+ gst_rtsp_stream_get_ssrc (stream, &ssrc);
+ gst_mikey_message_add_cs_srtp (msg, 0, ssrc, 0);
+ /* timestamp is now */
+ gst_mikey_message_add_t_now_ntp_utc (msg);
+ /* add some random data */
+ gst_mikey_message_add_rand_len (msg, 16);
+
+ /* the policy '0' is SRTP with the above discovered algorithms */
+ payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
+ gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
+
+ /* only AES-CM is supported */
+ byte = 1;
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1,
+ &byte);
+ /* Encryption key length */
+ byte = enc_key_length_from_cipher_name (srtpcipher);
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
+ &byte);
+ /* only HMAC-SHA1 */
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
+ &byte);
+ /* Authentication key length */
+ byte = auth_key_length_from_auth_name (srtpauth);
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
+ &byte);
+ /* we enable encryption on RTP and RTCP */
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
+ &byte);
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
+ &byte);
+ /* we enable authentication on RTP and RTCP */
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
+ &byte);
+ gst_mikey_message_add_payload (msg, payload);
+
+ /* make unencrypted KEMAC */
+ payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
+ gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL,
+ GST_MIKEY_MAC_NULL);
+
+ /* add the key in key data */
+ pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
+ gst_buffer_map (srtpkey, &info, GST_MAP_READ);
+ gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
+ info.data);
+ gst_buffer_unmap (srtpkey, &info);
+ /* add key data to KEMAC */
+ gst_mikey_payload_kemac_add_sub (payload, pkd);
+ gst_mikey_message_add_payload (msg, payload);
+
+ /* now serialize this to bytes */
+ bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
+ gst_mikey_message_unref (msg);
+ /* and make it into base64 */
+ data = g_bytes_get_data (bytes, &size);
+ base64 = g_base64_encode (data, size);
+ g_bytes_unref (bytes);
+
+ tmp = g_strdup_printf ("mikey %s", base64);
+ g_free (base64);
+
+ gst_sdp_media_add_attribute (smedia, "key-mgmt", tmp);
+ g_free (tmp);
+ } while (FALSE);
+
+ /* collect all other properties and add them to fmtp or attributes */
+ fmtp = g_string_new ("");
+ g_string_append_printf (fmtp, "%d ", caps_pt);
+ first = TRUE;
+ n_fields = gst_structure_n_fields (s);
+ for (j = 0; j < n_fields; j++) {
+ const gchar *fname, *fval;
+
+ fname = gst_structure_nth_field_name (s, j);
+
+ /* filter out standard properties */
+ if (!strcmp (fname, "media"))
+ continue;
+ if (!strcmp (fname, "payload"))
+ continue;
+ if (!strcmp (fname, "clock-rate"))
+ continue;
+ if (!strcmp (fname, "encoding-name"))
+ continue;
+ if (!strcmp (fname, "encoding-params"))
+ continue;
+ if (!strcmp (fname, "ssrc"))
+ continue;
+ if (!strcmp (fname, "clock-base"))
+ continue;
+ if (!strcmp (fname, "seqnum-base"))
+ continue;
+ if (g_str_has_prefix (fname, "srtp-"))
+ continue;
+ if (g_str_has_prefix (fname, "srtcp-"))
+ continue;
+
+ if (g_str_has_prefix (fname, "a-")) {
+ /* attribute */
+ if ((fval = gst_structure_get_string (s, fname)))
+ gst_sdp_media_add_attribute (smedia, fname + 2, fval);
+ continue;
+ }
+ if (g_str_has_prefix (fname, "x-")) {
+ /* attribute */
+ if ((fval = gst_structure_get_string (s, fname)))
+ gst_sdp_media_add_attribute (smedia, fname, fval);
+ continue;
+ }
+
+ if ((fval = gst_structure_get_string (s, fname))) {
+ g_string_append_printf (fmtp, "%s%s=%s", first ? "" : ";", fname, fval);
+ first = FALSE;
+ }
+ }
+ if (!first) {
+ tmp = g_string_free (fmtp, FALSE);
+ gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
+ g_free (tmp);
+ } else {
+ g_string_free (fmtp, TRUE);
+ }
+
+ update_sdp_from_tags (stream, smedia);
+
+ gst_sdp_message_add_media (sdp, smedia);
+ gst_sdp_media_free (smedia);
+
+ return;
+
+ /* ERRORS */
+no_multicast:
+ {
+ gst_sdp_media_free (smedia);
+ g_warning ("ignoring stream %d without multicast address",
+ gst_rtsp_stream_get_index (stream));
+ return;
+ }
+}
+
+/**
+ * gst_rtsp_sdp_from_media:
+ * @sdp: a #GstSDPMessage
+ * @info: (transfer none): a #GstSDPInfo
+ * @media: (transfer none): a #GstRTSPMedia
+ *
+ * Add @media specific info to @sdp. @info is used to configure the connection
+ * information in the SDP.
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
+ GstRTSPMedia * media)
+{
+ guint i, n_streams;
+ gchar *rangestr;
+
+ n_streams = gst_rtsp_media_n_streams (media);
+
+ rangestr = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ if (rangestr == NULL)
+ goto not_prepared;
+
+ gst_sdp_message_add_attribute (sdp, "range", rangestr);
+ g_free (rangestr);
+
+ for (i = 0; i < n_streams; i++) {
+ GstRTSPStream *stream;
+ GstCaps *caps;
+ GstStructure *s;
+ GstRTSPProfile profiles;
+ guint mask;
+
+ stream = gst_rtsp_media_get_stream (media, i);
+ caps = gst_rtsp_stream_get_caps (stream);
+
+ if (caps == NULL) {
+ g_warning ("ignoring stream %d without media type", i);
+ continue;
+ }
+
+ s = gst_caps_get_structure (caps, 0);
+ if (s == NULL) {
+ gst_caps_unref (caps);
+ g_warning ("ignoring stream %d without media type", i);
+ continue;
+ }
+
+ /* make a new media for each profile */
+ profiles = gst_rtsp_stream_get_profiles (stream);
+ mask = 1;
+ while (profiles >= mask) {
+ GstRTSPProfile prof = profiles & mask;
+
+ if (prof)
+ make_media (sdp, info, media, stream, s, prof);
+
+ mask <<= 1;
+ }
+ gst_caps_unref (caps);
+ }
+
+ {
+ GstNetTimeProvider *provider;
+
+ if ((provider =
+ gst_rtsp_media_get_time_provider (media, info->server_ip, 0))) {
+ GstClock *clock;
+ gchar *address, *str;
+ gint port;
+
+ g_object_get (provider, "clock", &clock, "address", &address, "port",
+ &port, NULL);
+
+ str = g_strdup_printf ("GstNetTimeProvider %s %s:%d %" G_GUINT64_FORMAT,
+ g_type_name (G_TYPE_FROM_INSTANCE (clock)), address, port,
+ gst_clock_get_time (clock));
+
+ gst_sdp_message_add_attribute (sdp, "x-gst-clock", str);
+ g_free (str);
+ gst_object_unref (clock);
+ g_free (address);
+ gst_object_unref (provider);
+ }
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+not_prepared:
+ {
+ GST_ERROR ("media %p is not prepared", media);
+ return FALSE;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/sdp/gstsdpmessage.h>
+
+#include "rtsp-media.h"
+
+#ifndef __GST_RTSP_SDP_H__
+#define __GST_RTSP_SDP_H__
+
+G_BEGIN_DECLS
+
+typedef struct {
+ gboolean is_ipv6;
+ const gchar *server_ip;
+} GstSDPInfo;
+
+/* creating SDP */
+gboolean gst_rtsp_sdp_from_media (GstSDPMessage *sdp, GstSDPInfo *info, GstRTSPMedia * media);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SDP_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-server
+ * @short_description: The main server object
+ * @see_also: #GstRTSPClient, #GstRTSPThreadPool
+ *
+ * The server object is the object listening for connections on a port and
+ * creating #GstRTSPClient objects to handle those connections.
+ *
+ * The server will listen on the address set with gst_rtsp_server_set_address()
+ * and the port or service configured with gst_rtsp_server_set_service().
+ * Use gst_rtsp_server_set_backlog() to configure the amount of pending requests
+ * that the server will keep. By default the server listens on the current
+ * network (0.0.0.0) and port 8554.
+ *
+ * The server will require an SSL connection when a TLS certificate has been
+ * set in the auth object with gst_rtsp_auth_set_tls_certificate().
+ *
+ * To start the server, use gst_rtsp_server_attach() to attach it to a
+ * #GMainContext. For more control, gst_rtsp_server_create_source() and
+ * gst_rtsp_server_create_socket() can be used to get a #GSource and #GSocket
+ * respectively.
+ *
+ * gst_rtsp_server_transfer_connection() can be used to transfer an existing
+ * socket to the RTSP server, for example from an HTTP server.
+ *
+ * Once the server socket is attached to a mainloop, it will start accepting
+ * connections. When a new connection is received, a new #GstRTSPClient object
+ * is created to handle the connection. The new client will be configured with
+ * the server #GstRTSPAuth, #GstRTSPMountPoints, #GstRTSPSessionPool and
+ * #GstRTSPThreadPool.
+ *
+ * The server uses the configured #GstRTSPThreadPool object to handle the
+ * remainder of the communication with this client.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#include <stdlib.h>
+#include <string.h>
+
+#include "rtsp-server-wfd.h"
+#include "rtsp-client-wfd.h"
+
+#define GST_RTSP_WFD_SERVER_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_WFD_SERVER, GstRTSPWFDServerPrivate))
+
+#define GST_RTSP_WFD_SERVER_GET_LOCK(server) (&(GST_RTSP_WFD_SERVER_CAST(server)->priv->lock))
+#define GST_RTSP_WFD_SERVER_LOCK(server) (g_mutex_lock(GST_RTSP_WFD_SERVER_GET_LOCK(server)))
+#define GST_RTSP_WFD_SERVER_UNLOCK(server) (g_mutex_unlock(GST_RTSP_WFD_SERVER_GET_LOCK(server)))
+
+struct _GstRTSPWFDServerPrivate
+{
+ GMutex lock; /* protects everything in this struct */
+
+ /* the clients that are connected */
+ GList *clients;
+ guint64 native_resolution;
+ guint64 supported_resolution;
+ guint8 audio_codec;
+};
+
+G_DEFINE_TYPE (GstRTSPWFDServer, gst_rtsp_wfd_server, GST_TYPE_RTSP_SERVER);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_wfd_server_debug);
+#define GST_CAT_DEFAULT rtsp_wfd_server_debug
+
+static void gst_rtsp_wfd_server_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_wfd_server_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_wfd_server_finalize (GObject * object);
+
+static GstRTSPClient *create_client_wfd (GstRTSPServer * server);
+static void client_connected_wfd (GstRTSPServer * server,
+ GstRTSPClient * client);
+
+static void
+gst_rtsp_wfd_server_class_init (GstRTSPWFDServerClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstRTSPServerClass *rtsp_server_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPWFDServerPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ rtsp_server_class = GST_RTSP_SERVER_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_wfd_server_get_property;
+ gobject_class->set_property = gst_rtsp_wfd_server_set_property;
+ gobject_class->finalize = gst_rtsp_wfd_server_finalize;
+
+ rtsp_server_class->create_client = create_client_wfd;
+ rtsp_server_class->client_connected = client_connected_wfd;
+
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_wfd_server_debug, "rtspwfdserver", 0,
+ "GstRTSPWFDServer");
+}
+
+static void
+gst_rtsp_wfd_server_init (GstRTSPWFDServer * server)
+{
+ GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE (server);
+
+ g_return_if_fail (priv != NULL);
+
+ server->priv = priv;
+ server->priv->native_resolution = 0;
+ server->priv->supported_resolution = 1;
+ server->priv->audio_codec = 2;
+ GST_INFO_OBJECT (server, "New server is initialized");
+}
+
+static void
+gst_rtsp_wfd_server_finalize (GObject * object)
+{
+ GstRTSPWFDServer *server = GST_RTSP_WFD_SERVER (object);
+ //GstRTSPWFDServerPrivate *priv = server->priv;
+
+ GST_DEBUG_OBJECT (server, "finalize server");
+
+ G_OBJECT_CLASS (gst_rtsp_wfd_server_parent_class)->finalize (object);
+}
+
+/**
+ * gst_rtsp_server_new:
+ *
+ * Create a new #GstRTSPWFDServer instance.
+ */
+GstRTSPWFDServer *
+gst_rtsp_wfd_server_new (void)
+{
+ GstRTSPWFDServer *result;
+
+ result = g_object_new (GST_TYPE_RTSP_WFD_SERVER, NULL);
+
+ return result;
+}
+
+static void
+gst_rtsp_wfd_server_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ //GstRTSPWFDServer *server = GST_RTSP_WFD_SERVER (object);
+
+ switch (propid) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_wfd_server_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ //GstRTSPWFDServer *server = GST_RTSP_WFD_SERVER (object);
+
+ switch (propid) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static gboolean
+_start_wfd (gpointer data)
+{
+ GstRTSPWFDClient *client = (GstRTSPWFDClient *) data;
+
+ GST_INFO_OBJECT (client, "WFD client is STARTing");
+
+ gst_rtsp_wfd_client_start_wfd (client);
+ return FALSE;
+}
+
+static void
+client_connected_wfd (GstRTSPServer * server, GstRTSPClient * client)
+{
+ GST_INFO_OBJECT (server, "Client is connected");
+
+ gst_rtsp_wfd_client_set_host_address (GST_RTSP_WFD_CLIENT_CAST (client),
+ gst_rtsp_server_get_address (server));
+ g_idle_add (_start_wfd, client);
+ return;
+}
+
+static GstRTSPClient *
+create_client_wfd (GstRTSPServer * server)
+{
+ GstRTSPWFDClient *client;
+ GstRTSPThreadPool *thread_pool = NULL;
+ GstRTSPSessionPool *session_pool = NULL;
+ GstRTSPMountPoints *mount_points = NULL;
+ GstRTSPAuth *auth = NULL;
+ GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE(server);
+
+ g_return_val_if_fail (priv != NULL, NULL);
+
+ GST_INFO_OBJECT (server, "New Client is being created");
+
+ /* a new client connected, create a session to handle the client. */
+ client = gst_rtsp_wfd_client_new ();
+
+ thread_pool = gst_rtsp_server_get_thread_pool (server);
+ session_pool = gst_rtsp_server_get_session_pool (server);
+ mount_points = gst_rtsp_server_get_mount_points (server);
+ auth = gst_rtsp_server_get_auth (server);
+
+ /* set the session pool that this client should use */
+ GST_RTSP_WFD_SERVER_LOCK (server);
+ gst_rtsp_client_set_session_pool (GST_RTSP_CLIENT_CAST (client),
+ session_pool);
+ /* set the mount points that this client should use */
+ gst_rtsp_client_set_mount_points (GST_RTSP_CLIENT_CAST (client),
+ mount_points);
+ /* set authentication manager */
+ gst_rtsp_client_set_auth (GST_RTSP_CLIENT_CAST (client), auth);
+ /* set threadpool */
+ gst_rtsp_client_set_thread_pool (GST_RTSP_CLIENT_CAST (client), thread_pool);
+
+ gst_rtsp_wfd_client_set_video_supported_resolution (client,
+ priv->supported_resolution);
+
+ gst_rtsp_wfd_client_set_video_native_resolution (client,
+ priv->native_resolution);
+
+ gst_rtsp_wfd_client_set_audio_codec (client,
+ priv->audio_codec);
+
+ GST_RTSP_WFD_SERVER_UNLOCK (server);
+
+ return GST_RTSP_CLIENT (client);
+}
+
+GstRTSPResult
+gst_rtsp_wfd_server_trigger_request (GstRTSPServer * server,
+ GstWFDTriggerType type)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GList *clients, *walk, *next;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), GST_RTSP_ERROR);
+
+ clients = gst_rtsp_server_client_filter (server, NULL, NULL);
+ if (clients == NULL) {
+ GST_ERROR_OBJECT (server, "There is no client in this server");
+ }
+
+ for (walk = clients; walk; walk = next) {
+ GstRTSPClient *client = walk->data;
+
+ next = g_list_next (walk);
+
+ res =
+ gst_rtsp_wfd_client_trigger_request (GST_RTSP_WFD_CLIENT (client),
+ type);
+ if (res != GST_RTSP_OK) {
+ GST_ERROR_OBJECT (server, "Failed to send trigger request %d", type);
+ }
+ g_object_unref (client);
+ }
+
+ return res;
+
+}
+
+GstRTSPResult
+gst_rtsp_wfd_server_set_supported_reso(GstRTSPWFDServer *server, guint64 supported_reso)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE(server);
+
+ g_return_val_if_fail (GST_IS_RTSP_WFD_SERVER (server), GST_RTSP_ERROR);
+ g_return_val_if_fail (priv != NULL, GST_RTSP_ERROR);
+
+ GST_RTSP_WFD_SERVER_LOCK (server);
+
+ priv->supported_resolution = supported_reso;
+
+ GST_RTSP_WFD_SERVER_UNLOCK (server);
+ return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_server_set_video_native_reso (GstRTSPWFDServer *server, guint64 native_reso)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE(server);
+
+ g_return_val_if_fail (GST_IS_RTSP_WFD_SERVER (server), GST_RTSP_ERROR);
+ g_return_val_if_fail (priv != NULL, GST_RTSP_ERROR);
+
+ GST_RTSP_WFD_SERVER_LOCK (server);
+
+ priv->native_resolution = native_reso;
+
+ GST_RTSP_WFD_SERVER_UNLOCK (server);
+ return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_server_set_audio_codec (GstRTSPWFDServer *server, guint8 audio_codec)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE(server);
+
+ g_return_val_if_fail (GST_IS_RTSP_WFD_SERVER (server), GST_RTSP_ERROR);
+ g_return_val_if_fail (priv != NULL, GST_RTSP_ERROR);
+
+ GST_RTSP_WFD_SERVER_LOCK (server);
+
+ priv->audio_codec = audio_codec;
+
+ GST_RTSP_WFD_SERVER_UNLOCK (server);
+ return res;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_SERVER_WFD_H__
+#define __GST_RTSP_SERVER_WFD_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPWFDServer GstRTSPWFDServer;
+typedef struct _GstRTSPWFDServerClass GstRTSPWFDServerClass;
+typedef struct _GstRTSPWFDServerPrivate GstRTSPWFDServerPrivate;
+
+#include "rtsp-session-pool.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-server.h"
+#include "rtsp-client-wfd.h"
+#include "rtsp-auth.h"
+
+#define GST_TYPE_RTSP_WFD_SERVER (gst_rtsp_wfd_server_get_type ())
+#define GST_IS_RTSP_WFD_SERVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_WFD_SERVER))
+#define GST_IS_RTSP_WFD_SERVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_WFD_SERVER))
+#define GST_RTSP_WFD_SERVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_WFD_SERVER, GstRTSPWFDServerClass))
+#define GST_RTSP_WFD_SERVER(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_WFD_SERVER, GstRTSPWFDServer))
+#define GST_RTSP_WFD_SERVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_WFD_SERVER, GstRTSPWFDServerClass))
+#define GST_RTSP_WFD_SERVER_CAST(obj) ((GstRTSPWFDServer*)(obj))
+#define GST_RTSP_WFD_SERVER_CLASS_CAST(klass) ((GstRTSPWFDServerClass*)(klass))
+
+/**
+ * GstRTSPWFDServer:
+ *
+ * This object listens on a port, creates and manages the clients connected to
+ * it.
+ */
+struct _GstRTSPWFDServer {
+ GstRTSPServer parent;
+
+ /*< private >*/
+ GstRTSPWFDServerPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPServerClass:
+ * @create_client: Create, configure a new GstRTSPClient
+ * object that handles the new connection on @socket. The default
+ * implementation will create a GstRTSPClient and will configure the
+ * mount-points, auth, session-pool and thread-pool on the client.
+ * @client_connected: emited when a new client connected.
+ *
+ * The RTSP server class structure
+ */
+struct _GstRTSPWFDServerClass {
+ GstRTSPServerClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType gst_rtsp_wfd_server_get_type (void);
+GstRTSPWFDServer * gst_rtsp_wfd_server_new (void);
+GstRTSPResult gst_rtsp_wfd_server_trigger_request (GstRTSPServer *server, GstWFDTriggerType type);
+
+GstRTSPResult gst_rtsp_wfd_server_set_supported_reso (GstRTSPWFDServer *server, guint64 supported_reso);
+GstRTSPResult gst_rtsp_wfd_server_set_video_native_reso (GstRTSPWFDServer *server, guint64 native_reso);
+GstRTSPResult gst_rtsp_wfd_server_set_audio_codec (GstRTSPWFDServer *server, guint8 audio_codec);
+
+#if 0
+void gst_rtsp_server_set_address (GstRTSPServer *server, const gchar *address);
+gchar * gst_rtsp_server_get_address (GstRTSPServer *server);
+
+void gst_rtsp_server_set_service (GstRTSPServer *server, const gchar *service);
+gchar * gst_rtsp_server_get_service (GstRTSPServer *server);
+
+void gst_rtsp_server_set_backlog (GstRTSPServer *server, gint backlog);
+gint gst_rtsp_server_get_backlog (GstRTSPServer *server);
+
+int gst_rtsp_server_get_bound_port (GstRTSPServer *server);
+
+void gst_rtsp_server_set_session_pool (GstRTSPServer *server, GstRTSPSessionPool *pool);
+GstRTSPSessionPool * gst_rtsp_server_get_session_pool (GstRTSPServer *server);
+
+void gst_rtsp_server_set_mount_points (GstRTSPServer *server, GstRTSPMountPoints *mounts);
+GstRTSPMountPoints * gst_rtsp_server_get_mount_points (GstRTSPServer *server);
+
+void gst_rtsp_server_set_auth (GstRTSPServer *server, GstRTSPAuth *auth);
+GstRTSPAuth * gst_rtsp_server_get_auth (GstRTSPServer *server);
+
+void gst_rtsp_server_set_thread_pool (GstRTSPServer *server, GstRTSPThreadPool *pool);
+GstRTSPThreadPool * gst_rtsp_server_get_thread_pool (GstRTSPServer *server);
+
+gboolean gst_rtsp_server_transfer_connection (GstRTSPServer * server, GSocket *socket,
+ const gchar * ip, gint port,
+ const gchar *initial_buffer);
+
+gboolean gst_rtsp_server_io_func (GSocket *socket, GIOCondition condition,
+ GstRTSPServer *server);
+
+GSocket * gst_rtsp_server_create_socket (GstRTSPServer *server,
+ GCancellable *cancellable,
+ GError **error);
+GSource * gst_rtsp_server_create_source (GstRTSPServer *server,
+ GCancellable * cancellable,
+ GError **error);
+guint gst_rtsp_server_attach (GstRTSPServer *server,
+ GMainContext *context);
+/**
+ * GstRTSPServerClientFilterFunc:
+ * @server: a #GstRTSPServer object
+ * @client: a #GstRTSPClient in @server
+ * @user_data: user data that has been given to gst_rtsp_server_client_filter()
+ *
+ * This function will be called by the gst_rtsp_server_client_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @client will be removed
+ * from @server.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @client untouched in
+ * @server.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @client to the result #GList of
+ * gst_rtsp_server_client_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPServerClientFilterFunc) (GstRTSPServer *server,
+ GstRTSPClient *client,
+ gpointer user_data);
+
+GList * gst_rtsp_server_client_filter (GstRTSPServer *server,
+ GstRTSPServerClientFilterFunc func,
+ gpointer user_data);
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SERVER_WFD_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-server
+ * @short_description: The main server object
+ * @see_also: #GstRTSPClient, #GstRTSPThreadPool
+ *
+ * The server object is the object listening for connections on a port and
+ * creating #GstRTSPClient objects to handle those connections.
+ *
+ * The server will listen on the address set with gst_rtsp_server_set_address()
+ * and the port or service configured with gst_rtsp_server_set_service().
+ * Use gst_rtsp_server_set_backlog() to configure the amount of pending requests
+ * that the server will keep. By default the server listens on the current
+ * network (0.0.0.0) and port 8554.
+ *
+ * The server will require an SSL connection when a TLS certificate has been
+ * set in the auth object with gst_rtsp_auth_set_tls_certificate().
+ *
+ * To start the server, use gst_rtsp_server_attach() to attach it to a
+ * #GMainContext. For more control, gst_rtsp_server_create_source() and
+ * gst_rtsp_server_create_socket() can be used to get a #GSource and #GSocket
+ * respectively.
+ *
+ * gst_rtsp_server_transfer_connection() can be used to transfer an existing
+ * socket to the RTSP server, for example from an HTTP server.
+ *
+ * Once the server socket is attached to a mainloop, it will start accepting
+ * connections. When a new connection is received, a new #GstRTSPClient object
+ * is created to handle the connection. The new client will be configured with
+ * the server #GstRTSPAuth, #GstRTSPMountPoints, #GstRTSPSessionPool and
+ * #GstRTSPThreadPool.
+ *
+ * The server uses the configured #GstRTSPThreadPool object to handle the
+ * remainder of the communication with this client.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#include <stdlib.h>
+#include <string.h>
+
+#include "rtsp-server.h"
+#include "rtsp-client.h"
+
+#define GST_RTSP_SERVER_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SERVER, GstRTSPServerPrivate))
+
+#define GST_RTSP_SERVER_GET_LOCK(server) (&(GST_RTSP_SERVER_CAST(server)->priv->lock))
+#define GST_RTSP_SERVER_LOCK(server) (g_mutex_lock(GST_RTSP_SERVER_GET_LOCK(server)))
+#define GST_RTSP_SERVER_UNLOCK(server) (g_mutex_unlock(GST_RTSP_SERVER_GET_LOCK(server)))
+
+struct _GstRTSPServerPrivate
+{
+ GMutex lock; /* protects everything in this struct */
+
+ /* server information */
+ gchar *address;
+ gchar *service;
+ gint backlog;
+
+ GSocket *socket;
+
+ /* sessions on this server */
+ GstRTSPSessionPool *session_pool;
+
+ /* mount points for this server */
+ GstRTSPMountPoints *mount_points;
+
+ /* authentication manager */
+ GstRTSPAuth *auth;
+
+ /* resource manager */
+ GstRTSPThreadPool *thread_pool;
+
+ /* the clients that are connected */
+ GList *clients;
+ guint clients_cookie;
+};
+
+#define DEFAULT_ADDRESS "0.0.0.0"
+#define DEFAULT_BOUND_PORT -1
+/* #define DEFAULT_ADDRESS "::0" */
+#define DEFAULT_SERVICE "8554"
+#define DEFAULT_BACKLOG 5
+
+/* Define to use the SO_LINGER option so that the server sockets can be resused
+ * sooner. Disabled for now because it is not very well implemented by various
+ * OSes and it causes clients to fail to read the TEARDOWN response. */
+#undef USE_SOLINGER
+
+enum
+{
+ PROP_0,
+ PROP_ADDRESS,
+ PROP_SERVICE,
+ PROP_BOUND_PORT,
+ PROP_BACKLOG,
+
+ PROP_SESSION_POOL,
+ PROP_MOUNT_POINTS,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_CLIENT_CONNECTED,
+ SIGNAL_LAST
+};
+
+G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_server_debug);
+#define GST_CAT_DEFAULT rtsp_server_debug
+
+typedef struct _ClientContext ClientContext;
+
+static guint gst_rtsp_server_signals[SIGNAL_LAST] = { 0 };
+
+static void gst_rtsp_server_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_server_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_server_finalize (GObject * object);
+
+static GstRTSPClient *default_create_client (GstRTSPServer * server);
+
+static void
+gst_rtsp_server_class_init (GstRTSPServerClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPServerPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_server_get_property;
+ gobject_class->set_property = gst_rtsp_server_set_property;
+ gobject_class->finalize = gst_rtsp_server_finalize;
+
+ /**
+ * GstRTSPServer::address:
+ *
+ * The address of the server. This is the address where the server will
+ * listen on.
+ */
+ g_object_class_install_property (gobject_class, PROP_ADDRESS,
+ g_param_spec_string ("address", "Address",
+ "The address the server uses to listen on", DEFAULT_ADDRESS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPServer::service:
+ *
+ * The service of the server. This is either a string with the service name or
+ * a port number (as a string) the server will listen on.
+ */
+ g_object_class_install_property (gobject_class, PROP_SERVICE,
+ g_param_spec_string ("service", "Service",
+ "The service or port number the server uses to listen on",
+ DEFAULT_SERVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPServer::bound-port:
+ *
+ * The actual port the server is listening on. Can be used to retrieve the
+ * port number when the server is started on port 0, which means bind to a
+ * random port. Set to -1 if the server has not been bound yet.
+ */
+ g_object_class_install_property (gobject_class, PROP_BOUND_PORT,
+ g_param_spec_int ("bound-port", "Bound port",
+ "The port number the server is listening on",
+ -1, G_MAXUINT16, DEFAULT_BOUND_PORT,
+ G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPServer::backlog:
+ *
+ * The backlog argument defines the maximum length to which the queue of
+ * pending connections for the server may grow. If a connection request arrives
+ * when the queue is full, the client may receive an error with an indication of
+ * ECONNREFUSED or, if the underlying protocol supports retransmission, the
+ * request may be ignored so that a later reattempt at connection succeeds.
+ */
+ g_object_class_install_property (gobject_class, PROP_BACKLOG,
+ g_param_spec_int ("backlog", "Backlog",
+ "The maximum length to which the queue "
+ "of pending connections may grow", 0, G_MAXINT, DEFAULT_BACKLOG,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPServer::session-pool:
+ *
+ * The session pool of the server. By default each server has a separate
+ * session pool but sessions can be shared between servers by setting the same
+ * session pool on multiple servers.
+ */
+ g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
+ g_param_spec_object ("session-pool", "Session Pool",
+ "The session pool to use for client session",
+ GST_TYPE_RTSP_SESSION_POOL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPServer::mount-points:
+ *
+ * The mount points to use for this server. By default the server has no
+ * mount points and thus cannot map urls to media streams.
+ */
+ g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
+ g_param_spec_object ("mount-points", "Mount Points",
+ "The mount points to use for client session",
+ GST_TYPE_RTSP_MOUNT_POINTS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED] =
+ g_signal_new ("client-connected", G_TYPE_FROM_CLASS (gobject_class),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPServerClass, client_connected),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CLIENT);
+
+ klass->create_client = default_create_client;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
+}
+
+static void
+gst_rtsp_server_init (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv = GST_RTSP_SERVER_GET_PRIVATE (server);
+
+ server->priv = priv;
+
+ g_mutex_init (&priv->lock);
+ priv->address = g_strdup (DEFAULT_ADDRESS);
+ priv->service = g_strdup (DEFAULT_SERVICE);
+ priv->socket = NULL;
+ priv->backlog = DEFAULT_BACKLOG;
+ priv->session_pool = gst_rtsp_session_pool_new ();
+ priv->mount_points = gst_rtsp_mount_points_new ();
+ priv->thread_pool = gst_rtsp_thread_pool_new ();
+}
+
+static void
+gst_rtsp_server_finalize (GObject * object)
+{
+ GstRTSPServer *server = GST_RTSP_SERVER (object);
+ GstRTSPServerPrivate *priv = server->priv;
+
+ GST_DEBUG_OBJECT (server, "finalize server");
+
+ g_free (priv->address);
+ g_free (priv->service);
+
+ if (priv->socket)
+ g_object_unref (priv->socket);
+
+ if (priv->session_pool)
+ g_object_unref (priv->session_pool);
+ if (priv->mount_points)
+ g_object_unref (priv->mount_points);
+ if (priv->thread_pool)
+ g_object_unref (priv->thread_pool);
+
+ if (priv->auth)
+ g_object_unref (priv->auth);
+
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_server_parent_class)->finalize (object);
+}
+
+/**
+ * gst_rtsp_server_new:
+ *
+ * Create a new #GstRTSPServer instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPServer
+ */
+GstRTSPServer *
+gst_rtsp_server_new (void)
+{
+ GstRTSPServer *result;
+
+ result = g_object_new (GST_TYPE_RTSP_SERVER, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_address:
+ * @server: a #GstRTSPServer
+ * @address: the address
+ *
+ * Configure @server to accept connections on the given address.
+ *
+ * This function must be called before the server is bound.
+ */
+void
+gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
+{
+ GstRTSPServerPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+ g_return_if_fail (address != NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ g_free (priv->address);
+ priv->address = g_strdup (address);
+ GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_address:
+ * @server: a #GstRTSPServer
+ *
+ * Get the address on which the server will accept connections.
+ *
+ * Returns: (transfer full): the server address. g_free() after usage.
+ */
+gchar *
+gst_rtsp_server_get_address (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ result = g_strdup (priv->address);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_get_bound_port:
+ * @server: a #GstRTSPServer
+ *
+ * Get the port number where the server was bound to.
+ *
+ * Returns: the port number
+ */
+int
+gst_rtsp_server_get_bound_port (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ GSocketAddress *address;
+ int result = -1;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), result);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ if (priv->socket == NULL)
+ goto out;
+
+ address = g_socket_get_local_address (priv->socket, NULL);
+ result = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (address));
+ g_object_unref (address);
+
+out:
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_service:
+ * @server: a #GstRTSPServer
+ * @service: the service
+ *
+ * Configure @server to accept connections on the given service.
+ * @service should be a string containing the service name (see services(5)) or
+ * a string containing a port number between 1 and 65535.
+ *
+ * When @service is set to "0", the server will listen on a random free
+ * port. The actual used port can be retrieved with
+ * gst_rtsp_server_get_bound_port().
+ *
+ * This function must be called before the server is bound.
+ */
+void
+gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
+{
+ GstRTSPServerPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+ g_return_if_fail (service != NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ g_free (priv->service);
+ priv->service = g_strdup (service);
+ GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_service:
+ * @server: a #GstRTSPServer
+ *
+ * Get the service on which the server will accept connections.
+ *
+ * Returns: (transfer full): the service. use g_free() after usage.
+ */
+gchar *
+gst_rtsp_server_get_service (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ result = g_strdup (priv->service);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_backlog:
+ * @server: a #GstRTSPServer
+ * @backlog: the backlog
+ *
+ * configure the maximum amount of requests that may be queued for the
+ * server.
+ *
+ * This function must be called before the server is bound.
+ */
+void
+gst_rtsp_server_set_backlog (GstRTSPServer * server, gint backlog)
+{
+ GstRTSPServerPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ priv->backlog = backlog;
+ GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_backlog:
+ * @server: a #GstRTSPServer
+ *
+ * The maximum amount of queued requests for the server.
+ *
+ * Returns: the server backlog.
+ */
+gint
+gst_rtsp_server_get_backlog (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ gint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ result = priv->backlog;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_session_pool:
+ * @server: a #GstRTSPServer
+ * @pool: (transfer none): a #GstRTSPSessionPool
+ *
+ * configure @pool to be used as the session pool of @server.
+ */
+void
+gst_rtsp_server_set_session_pool (GstRTSPServer * server,
+ GstRTSPSessionPool * pool)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPSessionPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ if (pool)
+ g_object_ref (pool);
+
+ GST_RTSP_SERVER_LOCK (server);
+ old = priv->session_pool;
+ priv->session_pool = pool;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_server_get_session_pool:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPSessionPool used as the session pool of @server.
+ *
+ * Returns: (transfer full): the #GstRTSPSessionPool used for sessions. g_object_unref() after
+ * usage.
+ */
+GstRTSPSessionPool *
+gst_rtsp_server_get_session_pool (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPSessionPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ if ((result = priv->session_pool))
+ g_object_ref (result);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_mount_points:
+ * @server: a #GstRTSPServer
+ * @mounts: (transfer none): a #GstRTSPMountPoints
+ *
+ * configure @mounts to be used as the mount points of @server.
+ */
+void
+gst_rtsp_server_set_mount_points (GstRTSPServer * server,
+ GstRTSPMountPoints * mounts)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPMountPoints *old;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ if (mounts)
+ g_object_ref (mounts);
+
+ GST_RTSP_SERVER_LOCK (server);
+ old = priv->mount_points;
+ priv->mount_points = mounts;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (old)
+ g_object_unref (old);
+}
+
+
+/**
+ * gst_rtsp_server_get_mount_points:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPMountPoints used as the mount points of @server.
+ *
+ * Returns: (transfer full): the #GstRTSPMountPoints of @server. g_object_unref() after
+ * usage.
+ */
+GstRTSPMountPoints *
+gst_rtsp_server_get_mount_points (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPMountPoints *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ if ((result = priv->mount_points))
+ g_object_ref (result);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_auth:
+ * @server: a #GstRTSPServer
+ * @auth: (transfer none): a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @server.
+ */
+void
+gst_rtsp_server_set_auth (GstRTSPServer * server, GstRTSPAuth * auth)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPAuth *old;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ if (auth)
+ g_object_ref (auth);
+
+ GST_RTSP_SERVER_LOCK (server);
+ old = priv->auth;
+ priv->auth = auth;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (old)
+ g_object_unref (old);
+}
+
+
+/**
+ * gst_rtsp_server_get_auth:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @server.
+ *
+ * Returns: (transfer full): the #GstRTSPAuth of @server. g_object_unref() after
+ * usage.
+ */
+GstRTSPAuth *
+gst_rtsp_server_get_auth (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPAuth *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ if ((result = priv->auth))
+ g_object_ref (result);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_set_thread_pool:
+ * @server: a #GstRTSPServer
+ * @pool: (transfer none): a #GstRTSPThreadPool
+ *
+ * configure @pool to be used as the thread pool of @server.
+ */
+void
+gst_rtsp_server_set_thread_pool (GstRTSPServer * server,
+ GstRTSPThreadPool * pool)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPThreadPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ priv = server->priv;
+
+ if (pool)
+ g_object_ref (pool);
+
+ GST_RTSP_SERVER_LOCK (server);
+ old = priv->thread_pool;
+ priv->thread_pool = pool;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_server_get_thread_pool:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPThreadPool used as the thread pool of @server.
+ *
+ * Returns: (transfer full): the #GstRTSPThreadPool of @server. g_object_unref() after
+ * usage.
+ */
+GstRTSPThreadPool *
+gst_rtsp_server_get_thread_pool (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv;
+ GstRTSPThreadPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ if ((result = priv->thread_pool))
+ g_object_ref (result);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+static void
+gst_rtsp_server_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPServer *server = GST_RTSP_SERVER (object);
+
+ switch (propid) {
+ case PROP_ADDRESS:
+ g_value_take_string (value, gst_rtsp_server_get_address (server));
+ break;
+ case PROP_SERVICE:
+ g_value_take_string (value, gst_rtsp_server_get_service (server));
+ break;
+ case PROP_BOUND_PORT:
+ g_value_set_int (value, gst_rtsp_server_get_bound_port (server));
+ break;
+ case PROP_BACKLOG:
+ g_value_set_int (value, gst_rtsp_server_get_backlog (server));
+ break;
+ case PROP_SESSION_POOL:
+ g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
+ break;
+ case PROP_MOUNT_POINTS:
+ g_value_take_object (value, gst_rtsp_server_get_mount_points (server));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_server_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPServer *server = GST_RTSP_SERVER (object);
+
+ switch (propid) {
+ case PROP_ADDRESS:
+ gst_rtsp_server_set_address (server, g_value_get_string (value));
+ break;
+ case PROP_SERVICE:
+ gst_rtsp_server_set_service (server, g_value_get_string (value));
+ break;
+ case PROP_BACKLOG:
+ gst_rtsp_server_set_backlog (server, g_value_get_int (value));
+ break;
+ case PROP_SESSION_POOL:
+ gst_rtsp_server_set_session_pool (server, g_value_get_object (value));
+ break;
+ case PROP_MOUNT_POINTS:
+ gst_rtsp_server_set_mount_points (server, g_value_get_object (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_server_create_socket:
+ * @server: a #GstRTSPServer
+ * @cancellable: (allow-none): a #GCancellable
+ * @error: (out): a #GError
+ *
+ * Create a #GSocket for @server. The socket will listen on the
+ * configured service.
+ *
+ * Returns: (transfer full): the #GSocket for @server or %NULL when an error
+ * occurred.
+ */
+GSocket *
+gst_rtsp_server_create_socket (GstRTSPServer * server,
+ GCancellable * cancellable, GError ** error)
+{
+ GstRTSPServerPrivate *priv;
+ GSocketConnectable *conn;
+ GSocketAddressEnumerator *enumerator;
+ GSocket *socket = NULL;
+#ifdef USE_SOLINGER
+ struct linger linger;
+#endif
+ GError *sock_error = NULL;
+ GError *bind_error = NULL;
+ guint16 port;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ GST_RTSP_SERVER_LOCK (server);
+ GST_DEBUG_OBJECT (server, "getting address info of %s/%s", priv->address,
+ priv->service);
+
+ /* resolve the server IP address */
+ port = atoi (priv->service);
+ if (port != 0 || !strcmp (priv->service, "0"))
+ conn = g_network_address_new (priv->address, port);
+ else
+ conn = g_network_service_new (priv->service, "tcp", priv->address);
+
+ enumerator = g_socket_connectable_enumerate (conn);
+ g_object_unref (conn);
+
+ /* create server socket, we loop through all the addresses until we manage to
+ * create a socket and bind. */
+ while (TRUE) {
+ GSocketAddress *sockaddr;
+
+ sockaddr =
+ g_socket_address_enumerator_next (enumerator, cancellable, error);
+ if (!sockaddr) {
+ if (!*error)
+ GST_DEBUG_OBJECT (server, "no more addresses %s",
+ *error ? (*error)->message : "");
+ else
+ GST_DEBUG_OBJECT (server, "failed to retrieve next address %s",
+ (*error)->message);
+ break;
+ }
+
+ /* only keep the first error */
+ socket = g_socket_new (g_socket_address_get_family (sockaddr),
+ G_SOCKET_TYPE_STREAM, G_SOCKET_PROTOCOL_TCP,
+ sock_error ? NULL : &sock_error);
+
+ if (socket == NULL) {
+ GST_DEBUG_OBJECT (server, "failed to make socket (%s), try next",
+ sock_error->message);
+ g_object_unref (sockaddr);
+ continue;
+ }
+
+ if (g_socket_bind (socket, sockaddr, TRUE, bind_error ? NULL : &bind_error)) {
+ /* ask what port the socket has been bound to */
+ if (port == 0 || !strcmp (priv->service, "0")) {
+ GError *addr_error = NULL;
+
+ g_object_unref (sockaddr);
+ sockaddr = g_socket_get_local_address (socket, &addr_error);
+
+ if (addr_error != NULL) {
+ GST_DEBUG_OBJECT (server,
+ "failed to get the local address of a bound socket %s",
+ addr_error->message);
+ g_clear_error (&addr_error);
+ break;
+ }
+ port =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
+
+ if (port != 0) {
+ g_free (priv->service);
+ priv->service = g_strdup_printf ("%d", port);
+ } else {
+ GST_DEBUG_OBJECT (server, "failed to get the port of a bound socket");
+ }
+ }
+ g_object_unref (sockaddr);
+ break;
+ }
+
+ GST_DEBUG_OBJECT (server, "failed to bind socket (%s), try next",
+ bind_error->message);
+ g_object_unref (sockaddr);
+ g_object_unref (socket);
+ socket = NULL;
+ }
+ g_object_unref (enumerator);
+
+ if (socket == NULL)
+ goto no_socket;
+
+ g_clear_error (&sock_error);
+ g_clear_error (&bind_error);
+
+ GST_DEBUG_OBJECT (server, "opened sending server socket");
+
+ /* keep connection alive; avoids SIGPIPE during write */
+ g_socket_set_keepalive (socket, TRUE);
+
+#if 0
+#ifdef USE_SOLINGER
+ /* make sure socket is reset 5 seconds after close. This ensure that we can
+ * reuse the socket quickly while still having a chance to send data to the
+ * client. */
+ linger.l_onoff = 1;
+ linger.l_linger = 5;
+ if (setsockopt (sockfd, SOL_SOCKET, SO_LINGER,
+ (void *) &linger, sizeof (linger)) < 0)
+ goto linger_failed;
+#endif
+#endif
+
+ /* set the server socket to nonblocking */
+ g_socket_set_blocking (socket, FALSE);
+
+ /* set listen backlog */
+ g_socket_set_listen_backlog (socket, priv->backlog);
+
+ if (!g_socket_listen (socket, error))
+ goto listen_failed;
+
+ GST_DEBUG_OBJECT (server, "listening on server socket %p with queue of %d",
+ socket, priv->backlog);
+
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return socket;
+
+ /* ERRORS */
+no_socket:
+ {
+ GST_ERROR_OBJECT (server, "failed to create socket");
+ goto close_error;
+ }
+#if 0
+#ifdef USE_SOLINGER
+linger_failed:
+ {
+ GST_ERROR_OBJECT (server, "failed to no linger socket: %s",
+ g_strerror (errno));
+ goto close_error;
+ }
+#endif
+#endif
+listen_failed:
+ {
+ GST_ERROR_OBJECT (server, "failed to listen on socket: %s",
+ (*error)->message);
+ goto close_error;
+ }
+close_error:
+ {
+ if (socket)
+ g_object_unref (socket);
+
+ if (sock_error) {
+ if (error == NULL)
+ g_propagate_error (error, sock_error);
+ else
+ g_error_free (sock_error);
+ }
+ if (bind_error) {
+ if ((error == NULL) || (*error == NULL))
+ g_propagate_error (error, bind_error);
+ else
+ g_error_free (bind_error);
+ }
+ GST_RTSP_SERVER_UNLOCK (server);
+ return NULL;
+ }
+}
+
+struct _ClientContext
+{
+ GstRTSPServer *server;
+ GstRTSPThread *thread;
+ GstRTSPClient *client;
+};
+
+static gboolean
+free_client_context (ClientContext * ctx)
+{
+ GST_DEBUG ("free context %p", ctx);
+
+ GST_RTSP_SERVER_LOCK (ctx->server);
+ if (ctx->thread)
+ gst_rtsp_thread_stop (ctx->thread);
+ GST_RTSP_SERVER_UNLOCK (ctx->server);
+
+ g_object_unref (ctx->client);
+ g_object_unref (ctx->server);
+ g_slice_free (ClientContext, ctx);
+
+ return G_SOURCE_REMOVE;
+}
+
+static void
+unmanage_client (GstRTSPClient * client, ClientContext * ctx)
+{
+ GstRTSPServer *server = ctx->server;
+ GstRTSPServerPrivate *priv = server->priv;
+
+ GST_DEBUG_OBJECT (server, "unmanage client %p", client);
+
+ GST_RTSP_SERVER_LOCK (server);
+ priv->clients = g_list_remove (priv->clients, ctx);
+ priv->clients_cookie++;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (ctx->thread) {
+ GSource *src;
+
+ src = g_idle_source_new ();
+ g_source_set_callback (src, (GSourceFunc) free_client_context, ctx, NULL);
+ g_source_attach (src, ctx->thread->context);
+ g_source_unref (src);
+ } else {
+ free_client_context (ctx);
+ }
+}
+
+/* add the client context to the active list of clients, takes ownership
+ * of client */
+static void
+manage_client (GstRTSPServer * server, GstRTSPClient * client)
+{
+ ClientContext *cctx;
+ GstRTSPServerPrivate *priv = server->priv;
+ GMainContext *mainctx = NULL;
+ GstRTSPContext ctx = { NULL };
+
+ GST_DEBUG_OBJECT (server, "manage client %p", client);
+
+ g_signal_emit (server, gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED], 0,
+ client);
+
+ cctx = g_slice_new0 (ClientContext);
+ cctx->server = g_object_ref (server);
+ cctx->client = client;
+
+ GST_RTSP_SERVER_LOCK (server);
+
+ ctx.server = server;
+ ctx.client = client;
+
+ cctx->thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+ GST_RTSP_THREAD_TYPE_CLIENT, &ctx);
+ if (cctx->thread)
+ mainctx = cctx->thread->context;
+ else {
+ GSource *source;
+ /* find the context to add the watch */
+ if ((source = g_main_current_source ()))
+ mainctx = g_source_get_context (source);
+ }
+
+ g_signal_connect (client, "closed", (GCallback) unmanage_client, cctx);
+ priv->clients = g_list_prepend (priv->clients, cctx);
+ priv->clients_cookie++;
+
+ gst_rtsp_client_attach (client, mainctx);
+
+ GST_RTSP_SERVER_UNLOCK (server);
+}
+
+static GstRTSPClient *
+default_create_client (GstRTSPServer * server)
+{
+ GstRTSPClient *client;
+ GstRTSPServerPrivate *priv = server->priv;
+
+ /* a new client connected, create a session to handle the client. */
+ client = gst_rtsp_client_new ();
+
+ /* set the session pool that this client should use */
+ GST_RTSP_SERVER_LOCK (server);
+ gst_rtsp_client_set_session_pool (client, priv->session_pool);
+ /* set the mount points that this client should use */
+ gst_rtsp_client_set_mount_points (client, priv->mount_points);
+ /* set authentication manager */
+ gst_rtsp_client_set_auth (client, priv->auth);
+ /* set threadpool */
+ gst_rtsp_client_set_thread_pool (client, priv->thread_pool);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return client;
+}
+
+/**
+ * gst_rtsp_server_transfer_connection:
+ * @server: a #GstRTSPServer
+ * @socket: (transfer full): a network socket
+ * @ip: the IP address of the remote client
+ * @port: the port used by the other end
+ * @initial_buffer: any initial data that was already read from the socket
+ *
+ * Take an existing network socket and use it for an RTSP connection. This
+ * is used when transferring a socket from an HTTP server which should be used
+ * as an RTSP over HTTP tunnel. The @initial_buffer contains any remaining data
+ * that the HTTP server read from the socket while parsing the HTTP header.
+ *
+ * Returns: TRUE if all was ok, FALSE if an error occurred.
+ */
+gboolean
+gst_rtsp_server_transfer_connection (GstRTSPServer * server, GSocket * socket,
+ const gchar * ip, gint port, const gchar * initial_buffer)
+{
+ GstRTSPClient *client = NULL;
+ GstRTSPServerClass *klass;
+ GstRTSPConnection *conn;
+ GstRTSPResult res;
+
+ klass = GST_RTSP_SERVER_GET_CLASS (server);
+
+ if (klass->create_client)
+ client = klass->create_client (server);
+ if (client == NULL)
+ goto client_failed;
+
+ GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
+ initial_buffer, &conn), no_connection);
+ g_object_unref (socket);
+
+ /* set connection on the client now */
+ gst_rtsp_client_set_connection (client, conn);
+
+ /* manage the client connection */
+ manage_client (server, client);
+
+ return TRUE;
+
+ /* ERRORS */
+client_failed:
+ {
+ GST_ERROR_OBJECT (server, "failed to create a client");
+ g_object_unref (socket);
+ return FALSE;
+ }
+no_connection:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+ GST_ERROR ("could not create connection from socket %p: %s", socket, str);
+ g_free (str);
+ g_object_unref (socket);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_server_io_func:
+ * @socket: a #GSocket
+ * @condition: the condition on @source
+ * @server: (transfer none): a #GstRTSPServer
+ *
+ * A default #GSocketSourceFunc that creates a new #GstRTSPClient to accept and handle a
+ * new connection on @socket or @server.
+ *
+ * Returns: TRUE if the source could be connected, FALSE if an error occurred.
+ */
+gboolean
+gst_rtsp_server_io_func (GSocket * socket, GIOCondition condition,
+ GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv = server->priv;
+ GstRTSPClient *client = NULL;
+ GstRTSPServerClass *klass;
+ GstRTSPResult res;
+ GstRTSPConnection *conn = NULL;
+ GstRTSPContext ctx = { NULL };
+
+ if (condition & G_IO_IN) {
+ /* a new client connected. */
+ GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, NULL),
+ accept_failed);
+
+ ctx.server = server;
+ ctx.conn = conn;
+ ctx.auth = priv->auth;
+ gst_rtsp_context_push_current (&ctx);
+
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_CONNECT))
+ goto connection_refused;
+
+ klass = GST_RTSP_SERVER_GET_CLASS (server);
+ /* a new client connected, create a client object to handle the client. */
+ if (klass->create_client)
+ client = klass->create_client (server);
+ if (client == NULL)
+ goto client_failed;
+
+ /* set connection on the client now */
+ gst_rtsp_client_set_connection (client, conn);
+
+ /* manage the client connection */
+ manage_client (server, client);
+ } else {
+ GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
+ }
+exit:
+ gst_rtsp_context_pop_current (&ctx);
+
+ return G_SOURCE_CONTINUE;
+
+ /* ERRORS */
+accept_failed:
+ {
+ gchar *str = gst_rtsp_strresult (res);
+ GST_ERROR_OBJECT (server, "Could not accept client on socket %p: %s",
+ socket, str);
+ g_free (str);
+ goto exit;
+ }
+connection_refused:
+ {
+ GST_ERROR_OBJECT (server, "connection refused");
+ gst_rtsp_connection_free (conn);
+ goto exit;
+ }
+client_failed:
+ {
+ GST_ERROR_OBJECT (server, "failed to create a client");
+ gst_rtsp_connection_free (conn);
+ goto exit;
+ }
+}
+
+static void
+watch_destroyed (GstRTSPServer * server)
+{
+ GstRTSPServerPrivate *priv = server->priv;
+
+ GST_DEBUG_OBJECT (server, "source destroyed");
+
+ g_object_unref (priv->socket);
+ priv->socket = NULL;
+ g_object_unref (server);
+}
+
+/**
+ * gst_rtsp_server_create_source:
+ * @server: a #GstRTSPServer
+ * @cancellable: (allow-none): a #GCancellable or %NULL.
+ * @error: (out): a #GError
+ *
+ * Create a #GSource for @server. The new source will have a default
+ * #GSocketSourceFunc of gst_rtsp_server_io_func().
+ *
+ * @cancellable if not %NULL can be used to cancel the source, which will cause
+ * the source to trigger, reporting the current condition (which is likely 0
+ * unless cancellation happened at the same time as a condition change). You can
+ * check for this in the callback using g_cancellable_is_cancelled().
+ *
+ * Returns: (transfer full): the #GSource for @server or %NULL when an error
+ * occurred. Free with g_source_unref ()
+ */
+GSource *
+gst_rtsp_server_create_source (GstRTSPServer * server,
+ GCancellable * cancellable, GError ** error)
+{
+ GstRTSPServerPrivate *priv;
+ GSocket *socket, *old;
+ GSource *source;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ socket = gst_rtsp_server_create_socket (server, NULL, error);
+ if (socket == NULL)
+ goto no_socket;
+
+ GST_RTSP_SERVER_LOCK (server);
+ old = priv->socket;
+ priv->socket = g_object_ref (socket);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (old)
+ g_object_unref (old);
+
+ /* create a watch for reads (new connections) and possible errors */
+ source = g_socket_create_source (socket, G_IO_IN |
+ G_IO_ERR | G_IO_HUP | G_IO_NVAL, cancellable);
+ g_object_unref (socket);
+
+ /* configure the callback */
+ g_source_set_callback (source,
+ (GSourceFunc) gst_rtsp_server_io_func, g_object_ref (server),
+ (GDestroyNotify) watch_destroyed);
+
+ return source;
+
+no_socket:
+ {
+ GST_ERROR_OBJECT (server, "failed to create socket");
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_server_attach:
+ * @server: a #GstRTSPServer
+ * @context: (allow-none): a #GMainContext
+ *
+ * Attaches @server to @context. When the mainloop for @context is run, the
+ * server will be dispatched. When @context is %NULL, the default context will be
+ * used).
+ *
+ * This function should be called when the server properties and urls are fully
+ * configured and the server is ready to start.
+ *
+ * Returns: the ID (greater than 0) for the source within the GMainContext.
+ */
+guint
+gst_rtsp_server_attach (GstRTSPServer * server, GMainContext * context)
+{
+ guint res;
+ GSource *source;
+ GError *error = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
+
+ source = gst_rtsp_server_create_source (server, NULL, &error);
+ if (source == NULL)
+ goto no_source;
+
+ res = g_source_attach (source, context);
+ g_source_unref (source);
+
+ return res;
+
+ /* ERRORS */
+no_source:
+ {
+ GST_ERROR_OBJECT (server, "failed to create watch: %s", error->message);
+ g_error_free (error);
+ return 0;
+ }
+}
+
+/**
+ * gst_rtsp_server_client_filter:
+ * @server: a #GstRTSPServer
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: user data passed to @func
+ *
+ * Call @func for each client managed by @server. The result value of @func
+ * determines what happens to the client. @func will be called with @server
+ * locked so no further actions on @server can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the client will be removed from
+ * @server.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the client will remain in @server.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the client will remain in @server but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each client.
+ *
+ * Returns: (element-type GstRTSPClient) (transfer full): a #GList with all
+ * clients for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_server_client_filter (GstRTSPServer * server,
+ GstRTSPServerClientFilterFunc func, gpointer user_data)
+{
+ GstRTSPServerPrivate *priv;
+ GList *result, *walk, *next;
+ GHashTable *visited;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+ priv = server->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ GST_RTSP_SERVER_LOCK (server);
+restart:
+ cookie = priv->clients_cookie;
+ for (walk = priv->clients; walk; walk = next) {
+ ClientContext *cctx = walk->data;
+ GstRTSPClient *client = cctx->client;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ next = g_list_next (walk);
+
+ if (func) {
+ /* only visit each media once */
+ if (g_hash_table_contains (visited, client))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (client));
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ res = func (server, client, user_data);
+
+ GST_RTSP_SERVER_LOCK (server);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->clients_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ gst_rtsp_client_close (client);
+
+ GST_RTSP_SERVER_LOCK (server);
+ changed |= (cookie != priv->clients_cookie);
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (client));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_SERVER_H__
+#define __GST_RTSP_SERVER_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPServer GstRTSPServer;
+typedef struct _GstRTSPServerClass GstRTSPServerClass;
+typedef struct _GstRTSPServerPrivate GstRTSPServerPrivate;
+
+#include "rtsp-session-pool.h"
+#include "rtsp-session.h"
+#include "rtsp-media.h"
+#include "rtsp-stream.h"
+#include "rtsp-stream-transport.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-client.h"
+#include "rtsp-context.h"
+#include "rtsp-server.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-media-factory.h"
+#include "rtsp-permissions.h"
+#include "rtsp-auth.h"
+#include "rtsp-token.h"
+#include "rtsp-session-media.h"
+#include "rtsp-sdp.h"
+#include "rtsp-media-factory-uri.h"
+#include "rtsp-params.h"
+
+#define GST_TYPE_RTSP_SERVER (gst_rtsp_server_get_type ())
+#define GST_IS_RTSP_SERVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SERVER))
+#define GST_IS_RTSP_SERVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SERVER))
+#define GST_RTSP_SERVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SERVER, GstRTSPServerClass))
+#define GST_RTSP_SERVER(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SERVER, GstRTSPServer))
+#define GST_RTSP_SERVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SERVER, GstRTSPServerClass))
+#define GST_RTSP_SERVER_CAST(obj) ((GstRTSPServer*)(obj))
+#define GST_RTSP_SERVER_CLASS_CAST(klass) ((GstRTSPServerClass*)(klass))
+
+/**
+ * GstRTSPServer:
+ *
+ * This object listens on a port, creates and manages the clients connected to
+ * it.
+ */
+struct _GstRTSPServer {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPServerPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPServerClass:
+ * @create_client: Create, configure a new GstRTSPClient
+ * object that handles the new connection on @socket. The default
+ * implementation will create a GstRTSPClient and will configure the
+ * mount-points, auth, session-pool and thread-pool on the client.
+ * @client_connected: emited when a new client connected.
+ *
+ * The RTSP server class structure
+ */
+struct _GstRTSPServerClass {
+ GObjectClass parent_class;
+
+ GstRTSPClient * (*create_client) (GstRTSPServer *server);
+
+ /* signals */
+ void (*client_connected) (GstRTSPServer *server, GstRTSPClient *client);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType gst_rtsp_server_get_type (void);
+
+GstRTSPServer * gst_rtsp_server_new (void);
+
+void gst_rtsp_server_set_address (GstRTSPServer *server, const gchar *address);
+gchar * gst_rtsp_server_get_address (GstRTSPServer *server);
+
+void gst_rtsp_server_set_service (GstRTSPServer *server, const gchar *service);
+gchar * gst_rtsp_server_get_service (GstRTSPServer *server);
+
+void gst_rtsp_server_set_backlog (GstRTSPServer *server, gint backlog);
+gint gst_rtsp_server_get_backlog (GstRTSPServer *server);
+
+int gst_rtsp_server_get_bound_port (GstRTSPServer *server);
+
+void gst_rtsp_server_set_session_pool (GstRTSPServer *server, GstRTSPSessionPool *pool);
+GstRTSPSessionPool * gst_rtsp_server_get_session_pool (GstRTSPServer *server);
+
+void gst_rtsp_server_set_mount_points (GstRTSPServer *server, GstRTSPMountPoints *mounts);
+GstRTSPMountPoints * gst_rtsp_server_get_mount_points (GstRTSPServer *server);
+
+void gst_rtsp_server_set_auth (GstRTSPServer *server, GstRTSPAuth *auth);
+GstRTSPAuth * gst_rtsp_server_get_auth (GstRTSPServer *server);
+
+void gst_rtsp_server_set_thread_pool (GstRTSPServer *server, GstRTSPThreadPool *pool);
+GstRTSPThreadPool * gst_rtsp_server_get_thread_pool (GstRTSPServer *server);
+
+gboolean gst_rtsp_server_transfer_connection (GstRTSPServer * server, GSocket *socket,
+ const gchar * ip, gint port,
+ const gchar *initial_buffer);
+
+gboolean gst_rtsp_server_io_func (GSocket *socket, GIOCondition condition,
+ GstRTSPServer *server);
+
+GSocket * gst_rtsp_server_create_socket (GstRTSPServer *server,
+ GCancellable *cancellable,
+ GError **error);
+GSource * gst_rtsp_server_create_source (GstRTSPServer *server,
+ GCancellable * cancellable,
+ GError **error);
+guint gst_rtsp_server_attach (GstRTSPServer *server,
+ GMainContext *context);
+
+/**
+ * GstRTSPServerClientFilterFunc:
+ * @server: a #GstRTSPServer object
+ * @client: a #GstRTSPClient in @server
+ * @user_data: user data that has been given to gst_rtsp_server_client_filter()
+ *
+ * This function will be called by the gst_rtsp_server_client_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @client will be removed
+ * from @server.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @client untouched in
+ * @server.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @client to the result #GList of
+ * gst_rtsp_server_client_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPServerClientFilterFunc) (GstRTSPServer *server,
+ GstRTSPClient *client,
+ gpointer user_data);
+
+GList * gst_rtsp_server_client_filter (GstRTSPServer *server,
+ GstRTSPServerClientFilterFunc func,
+ gpointer user_data);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SERVER_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-session-media
+ * @short_description: Media managed in a session
+ * @see_also: #GstRTSPMedia, #GstRTSPSession
+ *
+ * The #GstRTSPSessionMedia object manages a #GstRTSPMedia with a given path.
+ *
+ * With gst_rtsp_session_media_get_transport() and
+ * gst_rtsp_session_media_set_transport() the transports of a #GstRTSPStream of
+ * the managed #GstRTSPMedia can be retrieved and configured.
+ *
+ * Use gst_rtsp_session_media_set_state() to control the media state and
+ * transports.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+
+#include <string.h>
+
+#include "rtsp-session.h"
+
+#define GST_RTSP_SESSION_MEDIA_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaPrivate))
+
+struct _GstRTSPSessionMediaPrivate
+{
+ GMutex lock;
+ gchar *path; /* unmutable */
+ gint path_len; /* unmutable */
+ GstRTSPMedia *media; /* unmutable */
+ GstRTSPState state; /* protected by lock */
+ guint counter; /* protected by lock */
+
+ GPtrArray *transports; /* protected by lock */
+};
+
+enum
+{
+ PROP_0,
+ PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_media_debug);
+#define GST_CAT_DEFAULT rtsp_session_media_debug
+
+static void gst_rtsp_session_media_finalize (GObject * obj);
+
+G_DEFINE_TYPE (GstRTSPSessionMedia, gst_rtsp_session_media, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_session_media_class_init (GstRTSPSessionMediaClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPSessionMediaPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->finalize = gst_rtsp_session_media_finalize;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_session_media_debug, "rtspsessionmedia", 0,
+ "GstRTSPSessionMedia");
+}
+
+static void
+gst_rtsp_session_media_init (GstRTSPSessionMedia * media)
+{
+ GstRTSPSessionMediaPrivate *priv = GST_RTSP_SESSION_MEDIA_GET_PRIVATE (media);
+
+ media->priv = priv;
+
+ g_mutex_init (&priv->lock);
+ priv->state = GST_RTSP_STATE_INIT;
+}
+
+static void
+gst_rtsp_session_media_finalize (GObject * obj)
+{
+ GstRTSPSessionMedia *media;
+ GstRTSPSessionMediaPrivate *priv;
+
+ media = GST_RTSP_SESSION_MEDIA (obj);
+ priv = media->priv;
+
+ GST_INFO ("free session media %p", media);
+
+ gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
+
+ g_ptr_array_unref (priv->transports);
+
+ g_free (priv->path);
+ g_object_unref (priv->media);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_session_media_parent_class)->finalize (obj);
+}
+
+static void
+free_session_media (gpointer data)
+{
+ if (data)
+ g_object_unref (data);
+}
+
+/**
+ * gst_rtsp_session_media_new:
+ * @path: the path
+ * @media: (transfer full): the #GstRTSPMedia
+ *
+ * Create a new #GstRTSPSessionMedia that manages the streams
+ * in @media for @path. @media should be prepared.
+ *
+ * Ownership is taken of @media.
+ *
+ * Returns: (transfer full): a new #GstRTSPSessionMedia.
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_media_new (const gchar * path, GstRTSPMedia * media)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GstRTSPSessionMedia *result;
+ guint n_streams;
+ GstRTSPMediaStatus status;
+
+ g_return_val_if_fail (path != NULL, NULL);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ status = gst_rtsp_media_get_status (media);
+ g_return_val_if_fail (status == GST_RTSP_MEDIA_STATUS_PREPARED || status ==
+ GST_RTSP_MEDIA_STATUS_SUSPENDED, NULL);
+
+ result = g_object_new (GST_TYPE_RTSP_SESSION_MEDIA, NULL);
+ priv = result->priv;
+
+ priv->path = g_strdup (path);
+ priv->path_len = strlen (path);
+ priv->media = media;
+
+ /* prealloc the streams now, filled with NULL */
+ n_streams = gst_rtsp_media_n_streams (media);
+ priv->transports = g_ptr_array_new_full (n_streams, free_session_media);
+ g_ptr_array_set_size (priv->transports, n_streams);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_media_matches:
+ * @media: a #GstRTSPSessionMedia
+ * @path: a path
+ * @matched: (out): the amount of matched characters of @path
+ *
+ * Check if the path of @media matches @path. It @path matches, the amount of
+ * matched characters is returned in @matched.
+ *
+ * Returns: %TRUE when @path matches the path of @media.
+ */
+gboolean
+gst_rtsp_session_media_matches (GstRTSPSessionMedia * media,
+ const gchar * path, gint * matched)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ gint len;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
+ g_return_val_if_fail (path != NULL, FALSE);
+ g_return_val_if_fail (matched != NULL, FALSE);
+
+ priv = media->priv;
+ len = strlen (path);
+
+ /* path needs to be smaller than the media path */
+ if (len < priv->path_len)
+ return FALSE;
+
+ /* if media path is larger, it there should be a / following the path */
+ if (len > priv->path_len && path[priv->path_len] != '/')
+ return FALSE;
+
+ *matched = priv->path_len;
+
+ return strncmp (path, priv->path, priv->path_len) == 0;
+}
+
+/**
+ * gst_rtsp_session_media_get_media:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Get the #GstRTSPMedia that was used when constructing @media
+ *
+ * Returns: (transfer none): the #GstRTSPMedia of @media. Remains valid as long
+ * as @media is valid.
+ */
+GstRTSPMedia *
+gst_rtsp_session_media_get_media (GstRTSPSessionMedia * media)
+{
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+
+ return media->priv->media;
+}
+
+/**
+ * gst_rtsp_session_media_get_base_time:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Get the base_time of the #GstRTSPMedia in @media
+ *
+ * Returns: the base_time of the media.
+ */
+GstClockTime
+gst_rtsp_session_media_get_base_time (GstRTSPSessionMedia * media)
+{
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), GST_CLOCK_TIME_NONE);
+
+ return gst_rtsp_media_get_base_time (media->priv->media);
+}
+
+/**
+ * gst_rtsp_session_media_get_rtpinfo:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Retrieve the RTP-Info header string for all streams in @media
+ * with configured transports.
+ *
+ * Returns: (transfer full) (nullable): The RTP-Info as a string or
+ * %NULL when no RTP-Info could be generated, g_free() after usage.
+ */
+gchar *
+gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia * media)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GString *rtpinfo = NULL;
+ GstRTSPStreamTransport *transport;
+ GstRTSPStream *stream;
+ guint i, n_streams;
+ GstClockTime earliest = GST_CLOCK_TIME_NONE;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+
+ priv = media->priv;
+ g_mutex_lock (&priv->lock);
+
+ if (gst_rtsp_media_get_status (priv->media) != GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto not_prepared;
+
+ n_streams = priv->transports->len;
+
+ /* first step, take lowest running-time from all streams */
+ GST_LOG_OBJECT (media, "determining start time among %d transports",
+ n_streams);
+
+ for (i = 0; i < n_streams; i++) {
+ GstClockTime running_time;
+
+ transport = g_ptr_array_index (priv->transports, i);
+ if (transport == NULL) {
+ GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i);
+ continue;
+ }
+
+ stream = gst_rtsp_stream_transport_get_stream (transport);
+ if (!gst_rtsp_stream_get_rtpinfo (stream, NULL, NULL, NULL, &running_time))
+ continue;
+
+ GST_LOG_OBJECT (media, "running time of %d stream: %" GST_TIME_FORMAT, i,
+ GST_TIME_ARGS (running_time));
+
+ if (!GST_CLOCK_TIME_IS_VALID (earliest)) {
+ earliest = running_time;
+ } else {
+ earliest = MIN (earliest, running_time);
+ }
+ }
+
+ GST_LOG_OBJECT (media, "media start time: %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (earliest));
+
+ /* next step, scale all rtptime of all streams to lowest running-time */
+ GST_LOG_OBJECT (media, "collecting RTP info for %d transports", n_streams);
+
+ for (i = 0; i < n_streams; i++) {
+ gchar *stream_rtpinfo;
+
+ transport = g_ptr_array_index (priv->transports, i);
+ if (transport == NULL) {
+ GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i);
+ continue;
+ }
+
+ stream_rtpinfo =
+ gst_rtsp_stream_transport_get_rtpinfo (transport, earliest);
+ if (stream_rtpinfo == NULL) {
+ GST_DEBUG_OBJECT (media, "ignoring unknown RTPInfo %d", i);
+ continue;
+ }
+
+ if (rtpinfo == NULL)
+ rtpinfo = g_string_new ("");
+ else
+ g_string_append (rtpinfo, ", ");
+
+ g_string_append (rtpinfo, stream_rtpinfo);
+ g_free (stream_rtpinfo);
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (rtpinfo == NULL) {
+ GST_WARNING_OBJECT (media, "RTP info is empty");
+ return NULL;
+ }
+ return g_string_free (rtpinfo, FALSE);
+
+ /* ERRORS */
+not_prepared:
+ {
+ g_mutex_unlock (&priv->lock);
+ GST_ERROR_OBJECT (media, "media was not prepared");
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_session_media_set_transport:
+ * @media: a #GstRTSPSessionMedia
+ * @stream: a #GstRTSPStream
+ * @tr: (transfer full): a #GstRTSPTransport
+ *
+ * Configure the transport for @stream to @tr in @media.
+ *
+ * Returns: (transfer none): the new or updated #GstRTSPStreamTransport for @stream.
+ */
+GstRTSPStreamTransport *
+gst_rtsp_session_media_set_transport (GstRTSPSessionMedia * media,
+ GstRTSPStream * stream, GstRTSPTransport * tr)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GstRTSPStreamTransport *result;
+ guint idx;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (tr != NULL, NULL);
+ priv = media->priv;
+ idx = gst_rtsp_stream_get_index (stream);
+ g_return_val_if_fail (idx < priv->transports->len, NULL);
+
+ g_mutex_lock (&priv->lock);
+ result = g_ptr_array_index (priv->transports, idx);
+ if (result == NULL) {
+ result = gst_rtsp_stream_transport_new (stream, tr);
+ g_ptr_array_index (priv->transports, idx) = result;
+ g_mutex_unlock (&priv->lock);
+ } else {
+ gst_rtsp_stream_transport_set_transport (result, tr);
+ g_mutex_unlock (&priv->lock);
+ }
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_media_get_transport:
+ * @media: a #GstRTSPSessionMedia
+ * @idx: the stream index
+ *
+ * Get a previously created #GstRTSPStreamTransport for the stream at @idx.
+ *
+ * Returns: (transfer none): a #GstRTSPStreamTransport that is valid until the
+ * session of @media is unreffed.
+ */
+GstRTSPStreamTransport *
+gst_rtsp_session_media_get_transport (GstRTSPSessionMedia * media, guint idx)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GstRTSPStreamTransport *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+ priv = media->priv;
+ g_return_val_if_fail (idx < priv->transports->len, NULL);
+
+ g_mutex_lock (&priv->lock);
+ result = g_ptr_array_index (priv->transports, idx);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_media_alloc_channels:
+ * @media: a #GstRTSPSessionMedia
+ * @range: (out): a #GstRTSPRange
+ *
+ * Fill @range with the next available min and max channels for
+ * interleaved transport.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia * media,
+ GstRTSPRange * range)
+{
+ GstRTSPSessionMediaPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ range->min = priv->counter++;
+ range->max = priv->counter++;
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_session_media_set_state:
+ * @media: a #GstRTSPSessionMedia
+ * @state: the new state
+ *
+ * Tell the media object @media to change to @state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_session_media_set_state (GstRTSPSessionMedia * media, GstState state)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ gboolean ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ ret = gst_rtsp_media_set_state (priv->media, state, priv->transports);
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_session_media_set_rtsp_state:
+ * @media: a #GstRTSPSessionMedia
+ * @state: a #GstRTSPState
+ *
+ * Set the RTSP state of @media to @state.
+ */
+void
+gst_rtsp_session_media_set_rtsp_state (GstRTSPSessionMedia * media,
+ GstRTSPState state)
+{
+ GstRTSPSessionMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SESSION_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->state = state;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_media_get_rtsp_state:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Get the current RTSP state of @media.
+ *
+ * Returns: the current RTSP state of @media.
+ */
+GstRTSPState
+gst_rtsp_session_media_get_rtsp_state (GstRTSPSessionMedia * media)
+{
+ GstRTSPSessionMediaPrivate *priv;
+ GstRTSPState ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media),
+ GST_RTSP_STATE_INVALID);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ ret = priv->state;
+ g_mutex_unlock (&priv->lock);
+
+ return ret;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp/gstrtsptransport.h>
+
+#ifndef __GST_RTSP_SESSION_MEDIA_H__
+#define __GST_RTSP_SESSION_MEDIA_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_SESSION_MEDIA (gst_rtsp_session_media_get_type ())
+#define GST_IS_RTSP_SESSION_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION_MEDIA))
+#define GST_IS_RTSP_SESSION_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION_MEDIA))
+#define GST_RTSP_SESSION_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaClass))
+#define GST_RTSP_SESSION_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMedia))
+#define GST_RTSP_SESSION_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaClass))
+#define GST_RTSP_SESSION_MEDIA_CAST(obj) ((GstRTSPSessionMedia*)(obj))
+#define GST_RTSP_SESSION_MEDIA_CLASS_CAST(klass) ((GstRTSPSessionMediaClass*)(klass))
+
+typedef struct _GstRTSPSessionMedia GstRTSPSessionMedia;
+typedef struct _GstRTSPSessionMediaClass GstRTSPSessionMediaClass;
+typedef struct _GstRTSPSessionMediaPrivate GstRTSPSessionMediaPrivate;
+
+/**
+ * GstRTSPSessionMedia:
+ *
+ * State of a client session regarding a specific media identified by path.
+ */
+struct _GstRTSPSessionMedia
+{
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPSessionMediaPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstRTSPSessionMediaClass
+{
+ GObjectClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_session_media_get_type (void);
+
+GstRTSPSessionMedia * gst_rtsp_session_media_new (const gchar *path,
+ GstRTSPMedia *media);
+
+gboolean gst_rtsp_session_media_matches (GstRTSPSessionMedia *media,
+ const gchar *path,
+ gint * matched);
+GstRTSPMedia * gst_rtsp_session_media_get_media (GstRTSPSessionMedia *media);
+
+GstClockTime gst_rtsp_session_media_get_base_time (GstRTSPSessionMedia *media);
+/* control media */
+gboolean gst_rtsp_session_media_set_state (GstRTSPSessionMedia *media,
+ GstState state);
+
+void gst_rtsp_session_media_set_rtsp_state (GstRTSPSessionMedia *media,
+ GstRTSPState state);
+GstRTSPState gst_rtsp_session_media_get_rtsp_state (GstRTSPSessionMedia *media);
+
+/* get stream transport config */
+GstRTSPStreamTransport * gst_rtsp_session_media_set_transport (GstRTSPSessionMedia *media,
+ GstRTSPStream *stream,
+ GstRTSPTransport *tr);
+GstRTSPStreamTransport * gst_rtsp_session_media_get_transport (GstRTSPSessionMedia *media,
+ guint idx);
+
+gboolean gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia *media,
+ GstRTSPRange *range);
+
+gchar * gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia * media);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SESSION_MEDIA_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-session-pool
+ * @short_description: An object for managing sessions
+ * @see_also: #GstRTSPSession
+ *
+ * The #GstRTSPSessionPool object manages a list of #GstRTSPSession objects.
+ *
+ * The maximum number of sessions can be configured with
+ * gst_rtsp_session_pool_set_max_sessions(). The current number of sessions can
+ * be retrieved with gst_rtsp_session_pool_get_n_sessions().
+ *
+ * Use gst_rtsp_session_pool_create() to create a new #GstRTSPSession object.
+ * The session object can be found again with its id and
+ * gst_rtsp_session_pool_find().
+ *
+ * All sessions can be iterated with gst_rtsp_session_pool_filter().
+ *
+ * Run gst_rtsp_session_pool_cleanup() periodically to remove timed out sessions
+ * or use gst_rtsp_session_pool_create_watch() to be notified when session
+ * cleanup should be performed.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include "rtsp-session-pool.h"
+
+#define GST_RTSP_SESSION_POOL_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPoolPrivate))
+
+struct _GstRTSPSessionPoolPrivate
+{
+ GMutex lock; /* protects everything in this struct */
+ guint max_sessions;
+ GHashTable *sessions;
+ guint sessions_cookie;
+};
+
+#define DEFAULT_MAX_SESSIONS 0
+
+enum
+{
+ PROP_0,
+ PROP_MAX_SESSIONS,
+ PROP_LAST
+};
+
+static const gchar session_id_charset[] =
+ { 'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm', 'n', 'o',
+ 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z', 'A', 'B', 'C', 'D',
+ 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M', 'N', 'O', 'P', 'Q', 'R', 'S',
+ 'T', 'U', 'V', 'W', 'X', 'Y', 'Z', '0', '1', '2', '3', '4', '5', '6', '7',
+ '8', '9', '$', '-', '_', '.', '+'
+};
+
+enum
+{
+ SIGNAL_SESSION_REMOVED,
+ SIGNAL_LAST
+};
+
+static guint gst_rtsp_session_pool_signals[SIGNAL_LAST] = { 0 };
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_debug);
+#define GST_CAT_DEFAULT rtsp_session_debug
+
+static void gst_rtsp_session_pool_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_pool_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_pool_finalize (GObject * object);
+
+static gchar *create_session_id (GstRTSPSessionPool * pool);
+static GstRTSPSession *create_session (GstRTSPSessionPool * pool,
+ const gchar * id);
+
+G_DEFINE_TYPE (GstRTSPSessionPool, gst_rtsp_session_pool, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_session_pool_class_init (GstRTSPSessionPoolClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPSessionPoolPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_session_pool_get_property;
+ gobject_class->set_property = gst_rtsp_session_pool_set_property;
+ gobject_class->finalize = gst_rtsp_session_pool_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_MAX_SESSIONS,
+ g_param_spec_uint ("max-sessions", "Max Sessions",
+ "the maximum amount of sessions (0 = unlimited)",
+ 0, G_MAXUINT, DEFAULT_MAX_SESSIONS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED] =
+ g_signal_new ("session-removed", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPSessionPoolClass,
+ session_removed), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE,
+ 1, GST_TYPE_RTSP_SESSION);
+
+ klass->create_session_id = create_session_id;
+ klass->create_session = create_session;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsessionpool", 0,
+ "GstRTSPSessionPool");
+}
+
+static void
+gst_rtsp_session_pool_init (GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPoolPrivate *priv = GST_RTSP_SESSION_POOL_GET_PRIVATE (pool);
+
+ pool->priv = priv;
+
+ g_mutex_init (&priv->lock);
+ priv->sessions = g_hash_table_new_full (g_str_hash, g_str_equal,
+ NULL, g_object_unref);
+ priv->max_sessions = DEFAULT_MAX_SESSIONS;
+}
+
+static GstRTSPFilterResult
+remove_sessions_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
+ gpointer user_data)
+{
+ return GST_RTSP_FILTER_REMOVE;
+}
+
+static void
+gst_rtsp_session_pool_finalize (GObject * object)
+{
+ GstRTSPSessionPool *pool = GST_RTSP_SESSION_POOL (object);
+ GstRTSPSessionPoolPrivate *priv = pool->priv;
+
+ gst_rtsp_session_pool_filter (pool, remove_sessions_func, NULL);
+ g_hash_table_unref (priv->sessions);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_session_pool_parent_class)->finalize (object);
+}
+
+static void
+gst_rtsp_session_pool_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPSessionPool *pool = GST_RTSP_SESSION_POOL (object);
+
+ switch (propid) {
+ case PROP_MAX_SESSIONS:
+ g_value_set_uint (value, gst_rtsp_session_pool_get_max_sessions (pool));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtsp_session_pool_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPSessionPool *pool = GST_RTSP_SESSION_POOL (object);
+
+ switch (propid) {
+ case PROP_MAX_SESSIONS:
+ gst_rtsp_session_pool_set_max_sessions (pool, g_value_get_uint (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ break;
+ }
+}
+
+/**
+ * gst_rtsp_session_pool_new:
+ *
+ * Create a new #GstRTSPSessionPool instance.
+ *
+ * Returns: (transfer full): A new #GstRTSPSessionPool. g_object_unref() after
+ * usage.
+ */
+GstRTSPSessionPool *
+gst_rtsp_session_pool_new (void)
+{
+ GstRTSPSessionPool *result;
+
+ result = g_object_new (GST_TYPE_RTSP_SESSION_POOL, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_pool_set_max_sessions:
+ * @pool: a #GstRTSPSessionPool
+ * @max: the maximum number of sessions
+ *
+ * Configure the maximum allowed number of sessions in @pool to @max.
+ * A value of 0 means an unlimited amount of sessions.
+ */
+void
+gst_rtsp_session_pool_set_max_sessions (GstRTSPSessionPool * pool, guint max)
+{
+ GstRTSPSessionPoolPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SESSION_POOL (pool));
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->max_sessions = max;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_pool_get_max_sessions:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Get the maximum allowed number of sessions in @pool. 0 means an unlimited
+ * amount of sessions.
+ *
+ * Returns: the maximum allowed number of sessions.
+ */
+guint
+gst_rtsp_session_pool_get_max_sessions (GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ guint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), 0);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = priv->max_sessions;
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_pool_get_n_sessions:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Get the amount of active sessions in @pool.
+ *
+ * Returns: the amount of active sessions in @pool.
+ */
+guint
+gst_rtsp_session_pool_get_n_sessions (GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ guint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), 0);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = g_hash_table_size (priv->sessions);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_pool_find:
+ * @pool: the pool to search
+ * @sessionid: the session id
+ *
+ * Find the session with @sessionid in @pool. The access time of the session
+ * will be updated with gst_rtsp_session_touch().
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPSession with @sessionid
+ * or %NULL when the session did not exist. g_object_unref() after usage.
+ */
+GstRTSPSession *
+gst_rtsp_session_pool_find (GstRTSPSessionPool * pool, const gchar * sessionid)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ GstRTSPSession *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+ g_return_val_if_fail (sessionid != NULL, NULL);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = g_hash_table_lookup (priv->sessions, sessionid);
+ if (result) {
+ g_object_ref (result);
+ gst_rtsp_session_touch (result);
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+static gchar *
+create_session_id (GstRTSPSessionPool * pool)
+{
+ gchar id[17];
+ gint i;
+
+ for (i = 0; i < 16; i++) {
+ id[i] =
+ session_id_charset[g_random_int_range (0,
+ G_N_ELEMENTS (session_id_charset))];
+ }
+ id[16] = 0;
+
+ return g_uri_escape_string (id, NULL, FALSE);
+}
+
+static GstRTSPSession *
+create_session (GstRTSPSessionPool * pool, const gchar * id)
+{
+ return gst_rtsp_session_new (id);
+}
+
+/**
+ * gst_rtsp_session_pool_create:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Create a new #GstRTSPSession object in @pool.
+ *
+ * Returns: (transfer full): a new #GstRTSPSession.
+ */
+GstRTSPSession *
+gst_rtsp_session_pool_create (GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ GstRTSPSession *result = NULL;
+ GstRTSPSessionPoolClass *klass;
+ gchar *id = NULL;
+ guint retry;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+
+ priv = pool->priv;
+
+ klass = GST_RTSP_SESSION_POOL_GET_CLASS (pool);
+
+ retry = 0;
+ do {
+ /* start by creating a new random session id, we assume that this is random
+ * enough to not cause a collision, which we will check later */
+ if (klass->create_session_id)
+ id = klass->create_session_id (pool);
+ else
+ goto no_function;
+
+ if (id == NULL)
+ goto no_session;
+
+ g_mutex_lock (&priv->lock);
+ /* check session limit */
+ if (priv->max_sessions > 0) {
+ if (g_hash_table_size (priv->sessions) >= priv->max_sessions)
+ goto too_many_sessions;
+ }
+ /* check if the sessionid existed */
+ result = g_hash_table_lookup (priv->sessions, id);
+ if (result) {
+ /* found, retry with a different session id */
+ result = NULL;
+ retry++;
+ if (retry > 100)
+ goto collision;
+ } else {
+ /* not found, create session and insert it in the pool */
+ if (klass->create_session)
+ result = create_session (pool, id);
+ if (result == NULL)
+ goto too_many_sessions;
+ /* take additional ref for the pool */
+ g_object_ref (result);
+ g_hash_table_insert (priv->sessions,
+ (gchar *) gst_rtsp_session_get_sessionid (result), result);
+ priv->sessions_cookie++;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ g_free (id);
+ } while (result == NULL);
+
+ return result;
+
+ /* ERRORS */
+no_function:
+ {
+ GST_WARNING ("no create_session_id vmethod in GstRTSPSessionPool %p", pool);
+ return NULL;
+ }
+no_session:
+ {
+ GST_WARNING ("can't create session id with GstRTSPSessionPool %p", pool);
+ return NULL;
+ }
+collision:
+ {
+ GST_WARNING ("can't find unique sessionid for GstRTSPSessionPool %p", pool);
+ g_mutex_unlock (&priv->lock);
+ g_free (id);
+ return NULL;
+ }
+too_many_sessions:
+ {
+ GST_WARNING ("session pool reached max sessions of %d", priv->max_sessions);
+ g_mutex_unlock (&priv->lock);
+ g_free (id);
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_session_pool_remove:
+ * @pool: a #GstRTSPSessionPool
+ * @sess: (transfer none): a #GstRTSPSession
+ *
+ * Remove @sess from @pool, releasing the ref that the pool has on @sess.
+ *
+ * Returns: %TRUE if the session was found and removed.
+ */
+gboolean
+gst_rtsp_session_pool_remove (GstRTSPSessionPool * pool, GstRTSPSession * sess)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ gboolean found;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), FALSE);
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), FALSE);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_object_ref (sess);
+ found =
+ g_hash_table_remove (priv->sessions,
+ gst_rtsp_session_get_sessionid (sess));
+ if (found)
+ priv->sessions_cookie++;
+ g_mutex_unlock (&priv->lock);
+
+ if (found)
+ g_signal_emit (pool, gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED],
+ 0, sess);
+
+ g_object_unref (sess);
+
+ return found;
+}
+
+typedef struct
+{
+ GTimeVal now;
+ GstRTSPSessionPool *pool;
+ GList *removed;
+} CleanupData;
+
+static gboolean
+cleanup_func (gchar * sessionid, GstRTSPSession * sess, CleanupData * data)
+{
+ gboolean expired;
+
+ expired = gst_rtsp_session_is_expired (sess, &data->now);
+ if (expired) {
+ GST_DEBUG ("session expired");
+ data->removed = g_list_prepend (data->removed, g_object_ref (sess));
+ }
+ return expired;
+}
+
+/**
+ * gst_rtsp_session_pool_cleanup:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Inspect all the sessions in @pool and remove the sessions that are inactive
+ * for more than their timeout.
+ *
+ * Returns: the amount of sessions that got removed.
+ */
+guint
+gst_rtsp_session_pool_cleanup (GstRTSPSessionPool * pool)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ guint result;
+ CleanupData data;
+ GList *walk;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), 0);
+
+ priv = pool->priv;
+
+ g_get_current_time (&data.now);
+ data.pool = pool;
+ data.removed = NULL;
+
+ g_mutex_lock (&priv->lock);
+ result =
+ g_hash_table_foreach_remove (priv->sessions, (GHRFunc) cleanup_func,
+ &data);
+ if (result > 0)
+ priv->sessions_cookie++;
+ g_mutex_unlock (&priv->lock);
+
+ for (walk = data.removed; walk; walk = walk->next) {
+ GstRTSPSession *sess = walk->data;
+
+ g_signal_emit (pool,
+ gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED], 0, sess);
+
+ g_object_unref (sess);
+ }
+ g_list_free (data.removed);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_pool_filter:
+ * @pool: a #GstRTSPSessionPool
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each session in @pool. The result value of @func determines
+ * what happens to the session. @func will be called with the session pool
+ * locked so no further actions on @pool can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be set to the
+ * expired state with gst_rtsp_session_set_expired() and removed from
+ * @pool.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @pool.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @pool but
+ * will also be added with an additional ref to the result GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for all sessions.
+ *
+ * Returns: (element-type GstRTSPSession) (transfer full): a GList with all
+ * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_session_pool_filter (GstRTSPSessionPool * pool,
+ GstRTSPSessionPoolFilterFunc func, gpointer user_data)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ GHashTableIter iter;
+ gpointer key, value;
+ GList *result;
+ GHashTable *visited;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+
+ priv = pool->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ g_mutex_lock (&priv->lock);
+restart:
+ g_hash_table_iter_init (&iter, priv->sessions);
+ cookie = priv->sessions_cookie;
+ while (g_hash_table_iter_next (&iter, &key, &value)) {
+ GstRTSPSession *session = value;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ if (func) {
+ /* only visit each session once */
+ if (g_hash_table_contains (visited, session))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (session));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (pool, session, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->sessions_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ {
+ gboolean removed = TRUE;
+
+ if (changed)
+ /* something changed, check if we still have the session */
+ removed = g_hash_table_remove (priv->sessions, key);
+ else
+ g_hash_table_iter_remove (&iter);
+
+ if (removed) {
+ /* if we managed to remove the session, update the cookie and
+ * signal */
+ cookie = ++priv->sessions_cookie;
+ g_mutex_unlock (&priv->lock);
+
+ g_signal_emit (pool,
+ gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED], 0,
+ session);
+
+ g_mutex_lock (&priv->lock);
+ /* cookie could have changed again, make sure we restart */
+ changed |= (cookie != priv->sessions_cookie);
+ }
+ break;
+ }
+ case GST_RTSP_FILTER_REF:
+ /* keep ref */
+ result = g_list_prepend (result, g_object_ref (session));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
+
+typedef struct
+{
+ GSource source;
+ GstRTSPSessionPool *pool;
+ gint timeout;
+} GstPoolSource;
+
+static void
+collect_timeout (gchar * sessionid, GstRTSPSession * sess, GstPoolSource * psrc)
+{
+ gint timeout;
+ GTimeVal now;
+
+ g_get_current_time (&now);
+
+ timeout = gst_rtsp_session_next_timeout (sess, &now);
+ GST_INFO ("%p: next timeout: %d", sess, timeout);
+ if (psrc->timeout == -1 || timeout < psrc->timeout)
+ psrc->timeout = timeout;
+}
+
+static gboolean
+gst_pool_source_prepare (GSource * source, gint * timeout)
+{
+ GstRTSPSessionPoolPrivate *priv;
+ GstPoolSource *psrc;
+ gboolean result;
+
+ psrc = (GstPoolSource *) source;
+ psrc->timeout = -1;
+ priv = psrc->pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_hash_table_foreach (priv->sessions, (GHFunc) collect_timeout, psrc);
+ g_mutex_unlock (&priv->lock);
+
+ if (timeout)
+ *timeout = psrc->timeout;
+
+ result = psrc->timeout == 0;
+
+ GST_INFO ("prepare %d, %d", psrc->timeout, result);
+
+ return result;
+}
+
+static gboolean
+gst_pool_source_check (GSource * source)
+{
+ GST_INFO ("check");
+
+ return gst_pool_source_prepare (source, NULL);
+}
+
+static gboolean
+gst_pool_source_dispatch (GSource * source, GSourceFunc callback,
+ gpointer user_data)
+{
+ gboolean res;
+ GstPoolSource *psrc = (GstPoolSource *) source;
+ GstRTSPSessionPoolFunc func = (GstRTSPSessionPoolFunc) callback;
+
+ GST_INFO ("dispatch");
+
+ if (func)
+ res = func (psrc->pool, user_data);
+ else
+ res = FALSE;
+
+ return res;
+}
+
+static void
+gst_pool_source_finalize (GSource * source)
+{
+ GstPoolSource *psrc = (GstPoolSource *) source;
+
+ GST_INFO ("finalize %p", psrc);
+
+ g_object_unref (psrc->pool);
+ psrc->pool = NULL;
+}
+
+static GSourceFuncs gst_pool_source_funcs = {
+ gst_pool_source_prepare,
+ gst_pool_source_check,
+ gst_pool_source_dispatch,
+ gst_pool_source_finalize
+};
+
+/**
+ * gst_rtsp_session_pool_create_watch:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Create a #GSource that will be dispatched when the session should be cleaned
+ * up.
+ *
+ * Returns: (transfer full): a #GSource
+ */
+GSource *
+gst_rtsp_session_pool_create_watch (GstRTSPSessionPool * pool)
+{
+ GstPoolSource *source;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+
+ source = (GstPoolSource *) g_source_new (&gst_pool_source_funcs,
+ sizeof (GstPoolSource));
+ source->pool = g_object_ref (pool);
+
+ return (GSource *) source;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_SESSION_POOL_H__
+#define __GST_RTSP_SESSION_POOL_H__
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPSessionPool GstRTSPSessionPool;
+typedef struct _GstRTSPSessionPoolClass GstRTSPSessionPoolClass;
+typedef struct _GstRTSPSessionPoolPrivate GstRTSPSessionPoolPrivate;
+
+#include "rtsp-session.h"
+
+#define GST_TYPE_RTSP_SESSION_POOL (gst_rtsp_session_pool_get_type ())
+#define GST_IS_RTSP_SESSION_POOL(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION_POOL))
+#define GST_IS_RTSP_SESSION_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION_POOL))
+#define GST_RTSP_SESSION_POOL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPoolClass))
+#define GST_RTSP_SESSION_POOL(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPool))
+#define GST_RTSP_SESSION_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPoolClass))
+#define GST_RTSP_SESSION_POOL_CAST(obj) ((GstRTSPSessionPool*)(obj))
+#define GST_RTSP_SESSION_POOL_CLASS_CAST(klass) ((GstRTSPSessionPoolClass*)(klass))
+
+/**
+ * GstRTSPSessionPool:
+ *
+ * An object that keeps track of the active sessions. This object is usually
+ * attached to a #GstRTSPServer object to manage the sessions in that server.
+ */
+struct _GstRTSPSessionPool {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPSessionPoolPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPSessionPoolClass:
+ * @create_session_id: create a new random session id. Subclasses can create
+ * custom session ids and should not check if the session exists.
+ * @create_session: make a new session object.
+ * @session_removed: a session was removed from the pool
+ */
+struct _GstRTSPSessionPoolClass {
+ GObjectClass parent_class;
+
+ gchar * (*create_session_id) (GstRTSPSessionPool *pool);
+ GstRTSPSession * (*create_session) (GstRTSPSessionPool *pool, const gchar *id);
+
+ /* signals */
+ void (*session_removed) (GstRTSPSessionPool *pool,
+ GstRTSPSession *session);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE - 1];
+};
+
+/**
+ * GstRTSPSessionPoolFunc:
+ * @pool: a #GstRTSPSessionPool object
+ * @user_data: user data that has been given when registering the handler
+ *
+ * The function that will be called from the GSource watch on the session pool.
+ *
+ * The function will be called when the pool must be cleaned up because one or
+ * more sessions timed out.
+ *
+ * Returns: %FALSE if the source should be removed.
+ */
+typedef gboolean (*GstRTSPSessionPoolFunc) (GstRTSPSessionPool *pool, gpointer user_data);
+
+/**
+ * GstRTSPSessionPoolFilterFunc:
+ * @pool: a #GstRTSPSessionPool object
+ * @session: a #GstRTSPSession in @pool
+ * @user_data: user data that has been given to gst_rtsp_session_pool_filter()
+ *
+ * This function will be called by the gst_rtsp_session_pool_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @session will be removed
+ * from @pool.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @session untouched in
+ * @pool.
+ *
+ * A value of GST_RTSP_FILTER_REF will add @session to the result #GList of
+ * gst_rtsp_session_pool_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPSessionPoolFilterFunc) (GstRTSPSessionPool *pool,
+ GstRTSPSession *session,
+ gpointer user_data);
+
+
+GType gst_rtsp_session_pool_get_type (void);
+
+/* creating a session pool */
+GstRTSPSessionPool * gst_rtsp_session_pool_new (void);
+
+/* counting sessions */
+void gst_rtsp_session_pool_set_max_sessions (GstRTSPSessionPool *pool, guint max);
+guint gst_rtsp_session_pool_get_max_sessions (GstRTSPSessionPool *pool);
+
+guint gst_rtsp_session_pool_get_n_sessions (GstRTSPSessionPool *pool);
+
+/* managing sessions */
+GstRTSPSession * gst_rtsp_session_pool_create (GstRTSPSessionPool *pool);
+GstRTSPSession * gst_rtsp_session_pool_find (GstRTSPSessionPool *pool,
+ const gchar *sessionid);
+gboolean gst_rtsp_session_pool_remove (GstRTSPSessionPool *pool,
+ GstRTSPSession *sess);
+
+/* perform session maintenance */
+GList * gst_rtsp_session_pool_filter (GstRTSPSessionPool *pool,
+ GstRTSPSessionPoolFilterFunc func,
+ gpointer user_data);
+guint gst_rtsp_session_pool_cleanup (GstRTSPSessionPool *pool);
+GSource * gst_rtsp_session_pool_create_watch (GstRTSPSessionPool *pool);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SESSION_POOL_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-session
+ * @short_description: An object to manage media
+ * @see_also: #GstRTSPSessionPool, #GstRTSPSessionMedia, #GstRTSPMedia
+ *
+ * The #GstRTSPSession is identified by an id, unique in the
+ * #GstRTSPSessionPool that created the session and manages media and its
+ * configuration.
+ *
+ * A #GstRTSPSession has a timeout that can be retrieved with
+ * gst_rtsp_session_get_timeout(). You can check if the sessions is expired with
+ * gst_rtsp_session_is_expired(). gst_rtsp_session_touch() will reset the
+ * expiration counter of the session.
+ *
+ * When a client configures a media with SETUP, a session will be created to
+ * keep track of the configuration of that media. With
+ * gst_rtsp_session_manage_media(), the media is added to the managed media
+ * in the session. With gst_rtsp_session_release_media() the media can be
+ * released again from the session. Managed media is identified in the sessions
+ * with a url. Use gst_rtsp_session_get_media() to get the media that matches
+ * (part of) the given url.
+ *
+ * The media in a session can be iterated with gst_rtsp_session_filter().
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include <string.h>
+
+#include "rtsp-session.h"
+
+#define GST_RTSP_SESSION_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SESSION, GstRTSPSessionPrivate))
+
+struct _GstRTSPSessionPrivate
+{
+ GMutex lock; /* protects everything but sessionid and create_time */
+ gchar *sessionid;
+
+ guint timeout;
+ gboolean timeout_always_visible;
+ GTimeVal create_time; /* immutable */
+ GTimeVal last_access;
+ gint expire_count;
+
+ GList *medias;
+ guint medias_cookie;
+};
+
+#undef DEBUG
+
+#define DEFAULT_TIMEOUT 60
+#define DEFAULT_ALWAYS_VISIBLE FALSE
+
+enum
+{
+ PROP_0,
+ PROP_SESSIONID,
+ PROP_TIMEOUT,
+ PROP_TIMEOUT_ALWAYS_VISIBLE,
+ PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_debug);
+#define GST_CAT_DEFAULT rtsp_session_debug
+
+static void gst_rtsp_session_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_finalize (GObject * obj);
+
+G_DEFINE_TYPE (GstRTSPSession, gst_rtsp_session, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_session_class_init (GstRTSPSessionClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPSessionPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_session_get_property;
+ gobject_class->set_property = gst_rtsp_session_set_property;
+ gobject_class->finalize = gst_rtsp_session_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_SESSIONID,
+ g_param_spec_string ("sessionid", "Sessionid", "the session id",
+ NULL, G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY |
+ G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TIMEOUT,
+ g_param_spec_uint ("timeout", "timeout",
+ "the timeout of the session (0 = never)", 0, G_MAXUINT,
+ DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TIMEOUT_ALWAYS_VISIBLE,
+ g_param_spec_boolean ("timeout-always-visible", "Timeout Always Visible ",
+ "timeout always visible in header",
+ DEFAULT_ALWAYS_VISIBLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsession", 0,
+ "GstRTSPSession");
+}
+
+static void
+gst_rtsp_session_init (GstRTSPSession * session)
+{
+ GstRTSPSessionPrivate *priv = GST_RTSP_SESSION_GET_PRIVATE (session);
+
+ session->priv = priv;
+
+ GST_INFO ("init session %p", session);
+
+ g_mutex_init (&priv->lock);
+ priv->timeout = DEFAULT_TIMEOUT;
+ g_get_current_time (&priv->create_time);
+ gst_rtsp_session_touch (session);
+}
+
+static void
+gst_rtsp_session_finalize (GObject * obj)
+{
+ GstRTSPSession *session;
+ GstRTSPSessionPrivate *priv;
+
+ session = GST_RTSP_SESSION (obj);
+ priv = session->priv;
+
+ GST_INFO ("finalize session %p", session);
+
+ /* free all media */
+ g_list_free_full (priv->medias, g_object_unref);
+
+ /* free session id */
+ g_free (priv->sessionid);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_session_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_session_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPSession *session = GST_RTSP_SESSION (object);
+ GstRTSPSessionPrivate *priv = session->priv;
+
+ switch (propid) {
+ case PROP_SESSIONID:
+ g_value_set_string (value, priv->sessionid);
+ break;
+ case PROP_TIMEOUT:
+ g_value_set_uint (value, gst_rtsp_session_get_timeout (session));
+ break;
+ case PROP_TIMEOUT_ALWAYS_VISIBLE:
+ g_value_set_boolean (value, priv->timeout_always_visible);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_session_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPSession *session = GST_RTSP_SESSION (object);
+ GstRTSPSessionPrivate *priv = session->priv;
+
+ switch (propid) {
+ case PROP_SESSIONID:
+ g_free (priv->sessionid);
+ priv->sessionid = g_value_dup_string (value);
+ break;
+ case PROP_TIMEOUT:
+ gst_rtsp_session_set_timeout (session, g_value_get_uint (value));
+ break;
+ case PROP_TIMEOUT_ALWAYS_VISIBLE:
+ g_mutex_lock (&priv->lock);
+ priv->timeout_always_visible = g_value_get_boolean (value);
+ g_mutex_unlock (&priv->lock);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_session_manage_media:
+ * @sess: a #GstRTSPSession
+ * @path: the path for the media
+ * @media: (transfer full): a #GstRTSPMedia
+ *
+ * Manage the media object @obj in @sess. @path will be used to retrieve this
+ * media from the session with gst_rtsp_session_get_media().
+ *
+ * Ownership is taken from @media.
+ *
+ * Returns: (transfer none): a new @GstRTSPSessionMedia object.
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_manage_media (GstRTSPSession * sess, const gchar * path,
+ GstRTSPMedia * media)
+{
+ GstRTSPSessionPrivate *priv;
+ GstRTSPSessionMedia *result;
+ GstRTSPMediaStatus status;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
+ g_return_val_if_fail (path != NULL, NULL);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ status = gst_rtsp_media_get_status (media);
+ g_return_val_if_fail (status == GST_RTSP_MEDIA_STATUS_PREPARED || status ==
+ GST_RTSP_MEDIA_STATUS_SUSPENDED, NULL);
+
+ priv = sess->priv;
+
+ result = gst_rtsp_session_media_new (path, media);
+
+ g_mutex_lock (&priv->lock);
+ priv->medias = g_list_prepend (priv->medias, result);
+ priv->medias_cookie++;
+ g_mutex_unlock (&priv->lock);
+
+ GST_INFO ("manage new media %p in session %p", media, result);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_release_media:
+ * @sess: a #GstRTSPSession
+ * @media: (transfer none): a #GstRTSPMedia
+ *
+ * Release the managed @media in @sess, freeing the memory allocated by it.
+ *
+ * Returns: %TRUE if there are more media session left in @sess.
+ */
+gboolean
+gst_rtsp_session_release_media (GstRTSPSession * sess,
+ GstRTSPSessionMedia * media)
+{
+ GstRTSPSessionPrivate *priv;
+ GList *find;
+ gboolean more;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), FALSE);
+ g_return_val_if_fail (media != NULL, FALSE);
+
+ priv = sess->priv;
+
+ g_mutex_lock (&priv->lock);
+ find = g_list_find (priv->medias, media);
+ if (find) {
+ priv->medias = g_list_delete_link (priv->medias, find);
+ priv->medias_cookie++;
+ }
+ more = (priv->medias != NULL);
+ g_mutex_unlock (&priv->lock);
+
+ if (find)
+ g_object_unref (media);
+
+ return more;
+}
+
+/**
+ * gst_rtsp_session_get_media:
+ * @sess: a #GstRTSPSession
+ * @path: the path for the media
+ * @matched: (out): the amount of matched characters
+ *
+ * Get the session media for @path. @matched will contain the number of matched
+ * characters of @path.
+ *
+ * Returns: (transfer none): the configuration for @path in @sess.
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_get_media (GstRTSPSession * sess, const gchar * path,
+ gint * matched)
+{
+ GstRTSPSessionPrivate *priv;
+ GstRTSPSessionMedia *result;
+ GList *walk;
+ gint best;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
+ g_return_val_if_fail (path != NULL, NULL);
+
+ priv = sess->priv;
+ result = NULL;
+ best = 0;
+
+ g_mutex_lock (&priv->lock);
+ for (walk = priv->medias; walk; walk = g_list_next (walk)) {
+ GstRTSPSessionMedia *test;
+
+ test = (GstRTSPSessionMedia *) walk->data;
+
+ /* find largest match */
+ if (gst_rtsp_session_media_matches (test, path, matched)) {
+ if (best < *matched) {
+ result = test;
+ best = *matched;
+ }
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ *matched = best;
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_filter:
+ * @sess: a #GstRTSPSession
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each media in @sess. The result value of @func determines
+ * what happens to the media. @func will be called with @sess
+ * locked so no further actions on @sess can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the media will be removed from
+ * @sess.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the media will remain in @sess.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the media will remain in @sess but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for all media.
+ *
+ * Returns: (element-type GstRTSPSessionMedia) (transfer full): a GList with all
+ * media for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_session_filter (GstRTSPSession * sess,
+ GstRTSPSessionFilterFunc func, gpointer user_data)
+{
+ GstRTSPSessionPrivate *priv;
+ GList *result, *walk, *next;
+ GHashTable *visited;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
+
+ priv = sess->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ g_mutex_lock (&priv->lock);
+restart:
+ cookie = priv->medias_cookie;
+ for (walk = priv->medias; walk; walk = next) {
+ GstRTSPSessionMedia *media = walk->data;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ next = g_list_next (walk);
+
+ if (func) {
+ /* only visit each media once */
+ if (g_hash_table_contains (visited, media))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (media));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (sess, media, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->medias_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ if (changed)
+ priv->medias = g_list_remove (priv->medias, media);
+ else
+ priv->medias = g_list_delete_link (priv->medias, walk);
+ cookie = ++priv->medias_cookie;
+ g_object_unref (media);
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (media));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_new:
+ * @sessionid: a session id
+ *
+ * Create a new #GstRTSPSession instance with @sessionid.
+ *
+ * Returns: (transfer full): a new #GstRTSPSession
+ */
+GstRTSPSession *
+gst_rtsp_session_new (const gchar * sessionid)
+{
+ GstRTSPSession *result;
+
+ g_return_val_if_fail (sessionid != NULL, NULL);
+
+ result = g_object_new (GST_TYPE_RTSP_SESSION, "sessionid", sessionid, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_get_sessionid:
+ * @session: a #GstRTSPSession
+ *
+ * Get the sessionid of @session.
+ *
+ * Returns: (transfer none): the sessionid of @session. The value remains valid
+ * as long as @session is alive.
+ */
+const gchar *
+gst_rtsp_session_get_sessionid (GstRTSPSession * session)
+{
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (session), NULL);
+
+ return session->priv->sessionid;
+}
+
+/**
+ * gst_rtsp_session_get_header:
+ * @session: a #GstRTSPSession
+ *
+ * Get the string that can be placed in the Session header field.
+ *
+ * Returns: (transfer full): the Session header of @session. g_free() after usage.
+ */
+gchar *
+gst_rtsp_session_get_header (GstRTSPSession * session)
+{
+ GstRTSPSessionPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (session), NULL);
+
+ priv = session->priv;
+
+
+ g_mutex_lock (&priv->lock);
+ if (priv->timeout_always_visible || priv->timeout != 60)
+ result = g_strdup_printf ("%s; timeout=%d", priv->sessionid, priv->timeout);
+ else
+ result = g_strdup (priv->sessionid);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_session_set_timeout:
+ * @session: a #GstRTSPSession
+ * @timeout: the new timeout
+ *
+ * Configure @session for a timeout of @timeout seconds. The session will be
+ * cleaned up when there is no activity for @timeout seconds.
+ */
+void
+gst_rtsp_session_set_timeout (GstRTSPSession * session, guint timeout)
+{
+ GstRTSPSessionPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SESSION (session));
+
+ priv = session->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->timeout = timeout;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_get_timeout:
+ * @session: a #GstRTSPSession
+ *
+ * Get the timeout value of @session.
+ *
+ * Returns: the timeout of @session in seconds.
+ */
+guint
+gst_rtsp_session_get_timeout (GstRTSPSession * session)
+{
+ GstRTSPSessionPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (session), 0);
+
+ priv = session->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->timeout;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_session_touch:
+ * @session: a #GstRTSPSession
+ *
+ * Update the last_access time of the session to the current time.
+ */
+void
+gst_rtsp_session_touch (GstRTSPSession * session)
+{
+ GstRTSPSessionPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_SESSION (session));
+
+ priv = session->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_get_current_time (&priv->last_access);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_prevent_expire:
+ * @session: a #GstRTSPSession
+ *
+ * Prevent @session from expiring.
+ */
+void
+gst_rtsp_session_prevent_expire (GstRTSPSession * session)
+{
+ g_return_if_fail (GST_IS_RTSP_SESSION (session));
+
+ g_atomic_int_add (&session->priv->expire_count, 1);
+}
+
+/**
+ * gst_rtsp_session_allow_expire:
+ * @session: a #GstRTSPSession
+ *
+ * Allow @session to expire. This method must be called an equal
+ * amount of time as gst_rtsp_session_prevent_expire().
+ */
+void
+gst_rtsp_session_allow_expire (GstRTSPSession * session)
+{
+ g_atomic_int_add (&session->priv->expire_count, -1);
+}
+
+/**
+ * gst_rtsp_session_next_timeout:
+ * @session: a #GstRTSPSession
+ * @now: (transfer none): the current system time
+ *
+ * Get the amount of milliseconds till the session will expire.
+ *
+ * Returns: the amount of milliseconds since the session will time out.
+ */
+gint
+gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
+{
+ GstRTSPSessionPrivate *priv;
+ gint res;
+ GstClockTime last_access, now_ns;
+
+ g_return_val_if_fail (GST_IS_RTSP_SESSION (session), -1);
+ g_return_val_if_fail (now != NULL, -1);
+
+ priv = session->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (g_atomic_int_get (&priv->expire_count) != 0) {
+ /* touch session when the expire count is not 0 */
+ g_get_current_time (&priv->last_access);
+ }
+
+ last_access = GST_TIMEVAL_TO_TIME (priv->last_access);
+ /* add timeout allow for 5 seconds of extra time */
+ last_access += priv->timeout * GST_SECOND + (5 * GST_SECOND);
+ g_mutex_unlock (&priv->lock);
+
+ now_ns = GST_TIMEVAL_TO_TIME (*now);
+
+ if (last_access > now_ns)
+ res = GST_TIME_AS_MSECONDS (last_access - now_ns);
+ else
+ res = 0;
+
+ return res;
+}
+
+/**
+ * gst_rtsp_session_is_expired:
+ * @session: a #GstRTSPSession
+ * @now: (transfer none): the current system time
+ *
+ * Check if @session timeout out.
+ *
+ * Returns: %TRUE if @session timed out
+ */
+gboolean
+gst_rtsp_session_is_expired (GstRTSPSession * session, GTimeVal * now)
+{
+ gboolean res;
+
+ res = (gst_rtsp_session_next_timeout (session, now) == 0);
+
+ return res;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp/gstrtsptransport.h>
+
+#ifndef __GST_RTSP_SESSION_H__
+#define __GST_RTSP_SESSION_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_SESSION (gst_rtsp_session_get_type ())
+#define GST_IS_RTSP_SESSION(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION))
+#define GST_IS_RTSP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION))
+#define GST_RTSP_SESSION_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION, GstRTSPSessionClass))
+#define GST_RTSP_SESSION(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION, GstRTSPSession))
+#define GST_RTSP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION, GstRTSPSessionClass))
+#define GST_RTSP_SESSION_CAST(obj) ((GstRTSPSession*)(obj))
+#define GST_RTSP_SESSION_CLASS_CAST(klass) ((GstRTSPSessionClass*)(klass))
+
+typedef struct _GstRTSPSession GstRTSPSession;
+typedef struct _GstRTSPSessionClass GstRTSPSessionClass;
+typedef struct _GstRTSPSessionPrivate GstRTSPSessionPrivate;
+
+/**
+ * GstRTSPFilterResult:
+ * @GST_RTSP_FILTER_REMOVE: Remove session
+ * @GST_RTSP_FILTER_KEEP: Keep session in the pool
+ * @GST_RTSP_FILTER_REF: Ref session in the result list
+ *
+ * Possible return values for gst_rtsp_session_pool_filter().
+ */
+typedef enum
+{
+ GST_RTSP_FILTER_REMOVE,
+ GST_RTSP_FILTER_KEEP,
+ GST_RTSP_FILTER_REF,
+} GstRTSPFilterResult;
+
+#include "rtsp-media.h"
+#include "rtsp-session-media.h"
+
+/**
+ * GstRTSPSession:
+ *
+ * Session information kept by the server for a specific client.
+ * One client session, identified with a session id, can handle multiple medias
+ * identified with the url of a media.
+ */
+struct _GstRTSPSession {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPSessionPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstRTSPSessionClass {
+ GObjectClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_session_get_type (void);
+
+/* create a new session */
+GstRTSPSession * gst_rtsp_session_new (const gchar *sessionid);
+
+const gchar * gst_rtsp_session_get_sessionid (GstRTSPSession *session);
+
+gchar * gst_rtsp_session_get_header (GstRTSPSession *session);
+
+void gst_rtsp_session_set_timeout (GstRTSPSession *session, guint timeout);
+guint gst_rtsp_session_get_timeout (GstRTSPSession *session);
+
+/* session timeout stuff */
+void gst_rtsp_session_touch (GstRTSPSession *session);
+void gst_rtsp_session_prevent_expire (GstRTSPSession *session);
+void gst_rtsp_session_allow_expire (GstRTSPSession *session);
+gint gst_rtsp_session_next_timeout (GstRTSPSession *session, GTimeVal *now);
+gboolean gst_rtsp_session_is_expired (GstRTSPSession *session, GTimeVal *now);
+
+/* handle media in a session */
+GstRTSPSessionMedia * gst_rtsp_session_manage_media (GstRTSPSession *sess,
+ const gchar *path,
+ GstRTSPMedia *media);
+gboolean gst_rtsp_session_release_media (GstRTSPSession *sess,
+ GstRTSPSessionMedia *media);
+/* get media in a session */
+GstRTSPSessionMedia * gst_rtsp_session_get_media (GstRTSPSession *sess,
+ const gchar *path,
+ gint * matched);
+
+/**
+ * GstRTSPSessionFilterFunc:
+ * @sess: a #GstRTSPSession object
+ * @media: a #GstRTSPSessionMedia in @sess
+ * @user_data: user data that has been given to gst_rtsp_session_filter()
+ *
+ * This function will be called by the gst_rtsp_session_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @media will be removed
+ * from @sess.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @media untouched in
+ * @sess.
+ *
+ * A value of GST_RTSP_FILTER_REF will add @media to the result #GList of
+ * gst_rtsp_session_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPSessionFilterFunc) (GstRTSPSession *sess,
+ GstRTSPSessionMedia *media,
+ gpointer user_data);
+
+GList * gst_rtsp_session_filter (GstRTSPSession *sess,
+ GstRTSPSessionFilterFunc func,
+ gpointer user_data);
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SESSION_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-stream-transport
+ * @short_description: A media stream transport configuration
+ * @see_also: #GstRTSPStream, #GstRTSPSessionMedia
+ *
+ * The #GstRTSPStreamTransport configures the transport used by a
+ * #GstRTSPStream. It is usually manages by a #GstRTSPSessionMedia object.
+ *
+ * With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured
+ * to handle the RTP and RTCP packets from the stream, for example when they
+ * need to be sent over TCP.
+ *
+ * With gst_rtsp_stream_transport_set_active() the transports are added and
+ * removed from the stream.
+ *
+ * A #GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP
+ * is received from the client. It will also call
+ * gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+
+#include <string.h>
+#include <stdlib.h>
+
+#include "rtsp-stream-transport.h"
+
+#define GST_RTSP_STREAM_TRANSPORT_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportPrivate))
+
+struct _GstRTSPStreamTransportPrivate
+{
+ GstRTSPStream *stream;
+
+ GstRTSPSendFunc send_rtp;
+ GstRTSPSendFunc send_rtcp;
+ gpointer user_data;
+ GDestroyNotify notify;
+
+ GstRTSPKeepAliveFunc keep_alive;
+ gpointer ka_user_data;
+ GDestroyNotify ka_notify;
+ gboolean active;
+ gboolean timed_out;
+
+ GstRTSPTransport *transport;
+ GstRTSPUrl *url;
+
+ GObject *rtpsource;
+};
+
+enum
+{
+ PROP_0,
+ PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_stream_transport_debug);
+#define GST_CAT_DEFAULT rtsp_stream_transport_debug
+
+static void gst_rtsp_stream_transport_finalize (GObject * obj);
+
+G_DEFINE_TYPE (GstRTSPStreamTransport, gst_rtsp_stream_transport,
+ G_TYPE_OBJECT);
+
+static void
+gst_rtsp_stream_transport_class_init (GstRTSPStreamTransportClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPStreamTransportPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->finalize = gst_rtsp_stream_transport_finalize;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_stream_transport_debug, "rtspmediatransport",
+ 0, "GstRTSPStreamTransport");
+}
+
+static void
+gst_rtsp_stream_transport_init (GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamTransportPrivate *priv =
+ GST_RTSP_STREAM_TRANSPORT_GET_PRIVATE (trans);
+
+ trans->priv = priv;
+}
+
+static void
+gst_rtsp_stream_transport_finalize (GObject * obj)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ GstRTSPStreamTransport *trans;
+
+ trans = GST_RTSP_STREAM_TRANSPORT (obj);
+ priv = trans->priv;
+
+ /* remove callbacks now */
+ gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
+ gst_rtsp_stream_transport_set_keepalive (trans, NULL, NULL, NULL);
+
+ if (priv->transport)
+ gst_rtsp_transport_free (priv->transport);
+
+ if (priv->url)
+ gst_rtsp_url_free (priv->url);
+
+ G_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
+}
+
+/**
+ * gst_rtsp_stream_transport_new:
+ * @stream: a #GstRTSPStream
+ * @tr: (transfer full): a GstRTSPTransport
+ *
+ * Create a new #GstRTSPStreamTransport that can be used to manage
+ * @stream with transport @tr.
+ *
+ * Returns: (transfer full): a new #GstRTSPStreamTransport
+ */
+GstRTSPStreamTransport *
+gst_rtsp_stream_transport_new (GstRTSPStream * stream, GstRTSPTransport * tr)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ GstRTSPStreamTransport *trans;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (tr != NULL, NULL);
+
+ trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
+ priv = trans->priv;
+ priv->stream = stream;
+ priv->transport = tr;
+
+ return trans;
+}
+
+/**
+ * gst_rtsp_stream_transport_get_stream:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the #GstRTSPStream used when constructing @trans.
+ *
+ * Returns: (transfer none): the stream used when constructing @trans.
+ */
+GstRTSPStream *
+gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ return trans->priv->stream;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_callbacks:
+ * @trans: a #GstRTSPStreamTransport
+ * @send_rtp: (scope notified): a callback called when RTP should be sent
+ * @send_rtcp: (scope notified): a callback called when RTCP should be sent
+ * @user_data: (closure): user data passed to callbacks
+ * @notify: (allow-none): called with the user_data when no longer needed.
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
+ */
+void
+gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
+ GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
+ gpointer user_data, GDestroyNotify notify)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ priv->send_rtp = send_rtp;
+ priv->send_rtcp = send_rtcp;
+ if (priv->notify)
+ priv->notify (priv->user_data);
+ priv->user_data = user_data;
+ priv->notify = notify;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_keepalive:
+ * @trans: a #GstRTSPStreamTransport
+ * @keep_alive: (scope notified): a callback called when the receiver is active
+ * @user_data: (closure): user data passed to callback
+ * @notify: (allow-none): called with the user_data when no longer needed.
+ *
+ * Install callbacks that will be called when RTCP packets are received from the
+ * receiver of @trans.
+ */
+void
+gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
+ GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ priv->keep_alive = keep_alive;
+ if (priv->ka_notify)
+ priv->ka_notify (priv->ka_user_data);
+ priv->ka_user_data = user_data;
+ priv->ka_notify = notify;
+}
+
+
+/**
+ * gst_rtsp_stream_transport_set_transport:
+ * @trans: a #GstRTSPStreamTransport
+ * @tr: (transfer full): a client #GstRTSPTransport
+ *
+ * Set @tr as the client transport. This function takes ownership of the
+ * passed @tr.
+ */
+void
+gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
+ GstRTSPTransport * tr)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+ g_return_if_fail (tr != NULL);
+
+ priv = trans->priv;
+
+ /* keep track of the transports in the stream. */
+ if (priv->transport)
+ gst_rtsp_transport_free (priv->transport);
+ priv->transport = tr;
+}
+
+/**
+ * gst_rtsp_stream_transport_get_transport:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the transport configured in @trans.
+ *
+ * Returns: (transfer none): the transport configured in @trans. It remains
+ * valid for as long as @trans is valid.
+ */
+const GstRTSPTransport *
+gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ return trans->priv->transport;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_url:
+ * @trans: a #GstRTSPStreamTransport
+ * @url: (transfer none): a client #GstRTSPUrl
+ *
+ * Set @url as the client url.
+ */
+void
+gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans,
+ const GstRTSPUrl * url)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ priv = trans->priv;
+
+ /* keep track of the transports in the stream. */
+ if (priv->url)
+ gst_rtsp_url_free (priv->url);
+ priv->url = (url ? gst_rtsp_url_copy (url) : NULL);
+}
+
+/**
+ * gst_rtsp_stream_transport_get_url:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the url configured in @trans.
+ *
+ * Returns: (transfer none): the url configured in @trans. It remains
+ * valid for as long as @trans is valid.
+ */
+const GstRTSPUrl *
+gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ return trans->priv->url;
+}
+
+ /**
+ * gst_rtsp_stream_transport_get_rtpinfo:
+ * @trans: a #GstRTSPStreamTransport
+ * @start_time: a star time
+ *
+ * Get the RTP-Info string for @trans and @start_time.
+ *
+ * Returns: (transfer full) (nullable): the RTPInfo string for @trans
+ * and @start_time or %NULL when the RTP-Info could not be
+ * determined. g_free() after usage.
+ */
+gchar *
+gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport * trans,
+ GstClockTime start_time)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gchar *url_str;
+ GString *rtpinfo;
+ guint rtptime, seq, clock_rate;
+ GstClockTime running_time = GST_CLOCK_TIME_NONE;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+ priv = trans->priv;
+
+ if (!gst_rtsp_stream_get_rtpinfo (priv->stream, &rtptime, &seq, &clock_rate,
+ &running_time))
+ return NULL;
+
+ GST_DEBUG ("RTP time %u, seq %u, rate %u, running-time %" GST_TIME_FORMAT,
+ rtptime, seq, clock_rate, GST_TIME_ARGS (running_time));
+
+ if (GST_CLOCK_TIME_IS_VALID (running_time)
+ && GST_CLOCK_TIME_IS_VALID (start_time)) {
+ if (running_time > start_time) {
+ rtptime -=
+ gst_util_uint64_scale_int (running_time - start_time, clock_rate,
+ GST_SECOND);
+ } else {
+ rtptime +=
+ gst_util_uint64_scale_int (start_time - running_time, clock_rate,
+ GST_SECOND);
+ }
+ }
+ GST_DEBUG ("RTP time %u, for start-time %" GST_TIME_FORMAT,
+ rtptime, GST_TIME_ARGS (start_time));
+
+ rtpinfo = g_string_new ("");
+
+ url_str = gst_rtsp_url_get_request_uri (trans->priv->url);
+ g_string_append_printf (rtpinfo, "url=%s;seq=%u;rtptime=%u",
+ url_str, seq, rtptime);
+ g_free (url_str);
+
+ return g_string_free (rtpinfo, FALSE);
+}
+
+/**
+ * gst_rtsp_stream_transport_set_active:
+ * @trans: a #GstRTSPStreamTransport
+ * @active: new state of @trans
+ *
+ * Activate or deactivate datatransfer configured in @trans.
+ *
+ * Returns: %TRUE when the state was changed.
+ */
+gboolean
+gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport * trans,
+ gboolean active)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+
+ priv = trans->priv;
+
+ if (priv->active == active)
+ return FALSE;
+
+ if (active)
+ res = gst_rtsp_stream_add_transport (priv->stream, trans);
+ else
+ res = gst_rtsp_stream_remove_transport (priv->stream, trans);
+
+ if (res)
+ priv->active = active;
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_timed_out:
+ * @trans: a #GstRTSPStreamTransport
+ * @timedout: timed out value
+ *
+ * Set the timed out state of @trans to @timedout
+ */
+void
+gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport * trans,
+ gboolean timedout)
+{
+ g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+ trans->priv->timed_out = timedout;
+}
+
+/**
+ * gst_rtsp_stream_transport_is_timed_out:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Check if @trans is timed out.
+ *
+ * Returns: %TRUE if @trans timed out.
+ */
+gboolean
+gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport * trans)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+
+ return trans->priv->timed_out;
+}
+
+/**
+ * gst_rtsp_stream_transport_send_rtp:
+ * @trans: a #GstRTSPStreamTransport
+ * @buffer: (transfer none): a #GstBuffer
+ *
+ * Send @buffer to the installed RTP callback for @trans.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
+ GstBuffer * buffer)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gboolean res = FALSE;
+
+ priv = trans->priv;
+
+ if (priv->send_rtp)
+ res =
+ priv->send_rtp (buffer, priv->transport->interleaved.min,
+ priv->user_data);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_send_rtcp:
+ * @trans: a #GstRTSPStreamTransport
+ * @buffer: (transfer none): a #GstBuffer
+ *
+ * Send @buffer to the installed RTCP callback for @trans.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
+ GstBuffer * buffer)
+{
+ GstRTSPStreamTransportPrivate *priv;
+ gboolean res = FALSE;
+
+ priv = trans->priv;
+
+ if (priv->send_rtcp)
+ res =
+ priv->send_rtcp (buffer, priv->transport->interleaved.max,
+ priv->user_data);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_keep_alive:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Signal the installed keep_alive callback for @trans.
+ */
+void
+gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamTransportPrivate *priv;
+
+ priv = trans->priv;
+
+ if (priv->keep_alive)
+ priv->keep_alive (priv->ka_user_data);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtsprange.h>
+#include <gst/rtsp/gstrtspurl.h>
+
+#ifndef __GST_RTSP_STREAM_TRANSPORT_H__
+#define __GST_RTSP_STREAM_TRANSPORT_H__
+
+G_BEGIN_DECLS
+
+/* types for the media */
+#define GST_TYPE_RTSP_STREAM_TRANSPORT (gst_rtsp_stream_transport_get_type ())
+#define GST_IS_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT))
+#define GST_IS_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT))
+#define GST_RTSP_STREAM_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
+#define GST_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransport))
+#define GST_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
+#define GST_RTSP_STREAM_TRANSPORT_CAST(obj) ((GstRTSPStreamTransport*)(obj))
+#define GST_RTSP_STREAM_TRANSPORT_CLASS_CAST(klass) ((GstRTSPStreamTransportClass*)(klass))
+
+typedef struct _GstRTSPStreamTransport GstRTSPStreamTransport;
+typedef struct _GstRTSPStreamTransportClass GstRTSPStreamTransportClass;
+typedef struct _GstRTSPStreamTransportPrivate GstRTSPStreamTransportPrivate;
+
+#include "rtsp-stream.h"
+
+/**
+ * GstRTSPSendFunc:
+ * @buffer: a #GstBuffer
+ * @channel: a channel
+ * @user_data: user data
+ *
+ * Function registered with gst_rtsp_stream_transport_set_callbacks() and
+ * called when @buffer must be sent on @channel.
+ *
+ * Returns: %TRUE on success
+ */
+typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
+/**
+ * GstRTSPKeepAliveFunc:
+ * @user_data: user data
+ *
+ * Function registered with gst_rtsp_stream_transport_set_keepalive() and called
+ * when the stream is active.
+ */
+typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
+
+/**
+ * GstRTSPStreamTransport:
+ * @parent: parent instance
+ *
+ * A Transport description for a stream
+ */
+struct _GstRTSPStreamTransport {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPStreamTransportPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstRTSPStreamTransportClass {
+ GObjectClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_stream_transport_get_type (void);
+
+GstRTSPStreamTransport * gst_rtsp_stream_transport_new (GstRTSPStream *stream,
+ GstRTSPTransport *tr);
+
+GstRTSPStream * gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport *trans);
+
+void gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport *trans,
+ GstRTSPTransport * tr);
+const GstRTSPTransport * gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport *trans);
+
+void gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport *trans,
+ const GstRTSPUrl * url);
+const GstRTSPUrl * gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport *trans);
+
+
+gchar * gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport *trans,
+ GstClockTime start_time);
+
+void gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport *trans,
+ GstRTSPSendFunc send_rtp,
+ GstRTSPSendFunc send_rtcp,
+ gpointer user_data,
+ GDestroyNotify notify);
+void gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport *trans,
+ GstRTSPKeepAliveFunc keep_alive,
+ gpointer user_data,
+ GDestroyNotify notify);
+void gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport *trans);
+
+gboolean gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport *trans,
+ gboolean active);
+
+void gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport *trans,
+ gboolean timedout);
+gboolean gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport *trans);
+
+
+
+gboolean gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport *trans,
+ GstBuffer *buffer);
+gboolean gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport *trans,
+ GstBuffer *buffer);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_STREAM_TRANSPORT_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-stream
+ * @short_description: A media stream
+ * @see_also: #GstRTSPMedia
+ *
+ * The #GstRTSPStream object manages the data transport for one stream. It
+ * is created from a payloader element and a source pad that produce the RTP
+ * packets for the stream.
+ *
+ * With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
+ * and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
+ *
+ * The #GstRTSPStream will use the configured addresspool, as set with
+ * gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
+ * stream. With gst_rtsp_stream_get_multicast_address() you can get the
+ * configured address.
+ *
+ * With gst_rtsp_stream_get_server_port () you can get the port that the server
+ * will use to receive RTCP. This is the part that the clients will use to send
+ * RTCP to.
+ *
+ * With gst_rtsp_stream_add_transport() destinations can be added where the
+ * stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
+ * the destination again.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include <gio/gio.h>
+
+#include <gst/app/gstappsrc.h>
+#include <gst/app/gstappsink.h>
+
+#include "rtsp-stream.h"
+
+#define GST_RTSP_STREAM_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
+
+struct _GstRTSPStreamPrivate
+{
+ GMutex lock;
+ guint idx;
+ GstPad *srcpad;
+ GstElement *payloader;
+ guint buffer_size;
+ gboolean is_joined;
+ gchar *control;
+
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans protocols;
+
+ /* pads on the rtpbin */
+ GstPad *send_rtp_sink;
+ GstPad *recv_sink[2];
+ GstPad *send_src[2];
+
+ /* the RTPSession object */
+ GObject *session;
+
+ /* SRTP encoder/decoder */
+ GstElement *srtpenc;
+ GstElement *srtpdec;
+ GHashTable *keys;
+
+ /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
+ * sockets */
+ GstElement *udpsrc_v4[2];
+
+ /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
+ * sockets */
+ GstElement *udpsrc_v6[2];
+
+ GstElement *udpsink[2];
+
+ /* for TCP transport */
+ GstElement *appsrc[2];
+ GstElement *appqueue[2];
+ GstElement *appsink[2];
+
+ GstElement *tee[2];
+ GstElement *funnel[2];
+
+ /* server ports for sending/receiving over ipv4 */
+ GstRTSPRange server_port_v4;
+ GstRTSPAddress *server_addr_v4;
+ gboolean have_ipv4;
+
+ /* server ports for sending/receiving over ipv6 */
+ GstRTSPRange server_port_v6;
+ GstRTSPAddress *server_addr_v6;
+ gboolean have_ipv6;
+
+ /* multicast addresses */
+ GstRTSPAddressPool *pool;
+ GstRTSPAddress *addr_v4;
+ GstRTSPAddress *addr_v6;
+
+ /* the caps of the stream */
+ gulong caps_sig;
+ GstCaps *caps;
+
+ /* transports we stream to */
+ guint n_active;
+ GList *transports;
+ guint transports_cookie;
+ GList *tr_cache;
+ guint tr_cache_cookie;
+
+ gint dscp_qos;
+
+ /* stream blocking */
+ gulong blocked_id;
+ gboolean blocking;
+};
+
+#define DEFAULT_CONTROL NULL
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+ GST_RTSP_LOWER_TRANS_TCP
+
+enum
+{
+ PROP_0,
+ PROP_CONTROL,
+ PROP_PROFILES,
+ PROP_PROTOCOLS,
+ PROP_LAST
+};
+
+enum
+{
+ SIGNAL_NEW_RTP_ENCODER,
+ SIGNAL_NEW_RTCP_ENCODER,
+ SIGNAL_RTCP_STATS,
+ SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
+#define GST_CAT_DEFAULT rtsp_stream_debug
+
+static GQuark ssrc_stream_map_key;
+
+static void gst_rtsp_stream_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_stream_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+
+static void gst_rtsp_stream_finalize (GObject * obj);
+
+static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
+
+G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_stream_get_property;
+ gobject_class->set_property = gst_rtsp_stream_set_property;
+ gobject_class->finalize = gst_rtsp_stream_finalize;
+
+ g_object_class_install_property (gobject_class, PROP_CONTROL,
+ g_param_spec_string ("control", "Control",
+ "The control string for this stream", DEFAULT_CONTROL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
+ g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
+ g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+ gst_rtsp_stream_signals[SIGNAL_RTCP_STATS] =
+ g_signal_new ("rtcp-statistics", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_STRUCTURE);
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
+
+ ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
+}
+
+static void
+gst_rtsp_stream_init (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+
+ GST_DEBUG ("new stream %p", stream);
+
+ stream->priv = priv;
+
+ priv->dscp_qos = -1;
+ priv->control = g_strdup (DEFAULT_CONTROL);
+ priv->profiles = DEFAULT_PROFILES;
+ priv->protocols = DEFAULT_PROTOCOLS;
+
+ g_mutex_init (&priv->lock);
+
+ priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
+ NULL, (GDestroyNotify) gst_caps_unref);
+}
+
+static void
+gst_rtsp_stream_finalize (GObject * obj)
+{
+ GstRTSPStream *stream;
+ GstRTSPStreamPrivate *priv;
+
+ stream = GST_RTSP_STREAM (obj);
+ priv = stream->priv;
+
+ GST_DEBUG ("finalize stream %p", stream);
+
+ /* we really need to be unjoined now */
+ g_return_if_fail (!priv->is_joined);
+
+ if (priv->addr_v4)
+ gst_rtsp_address_free (priv->addr_v4);
+ if (priv->addr_v6)
+ gst_rtsp_address_free (priv->addr_v6);
+ if (priv->server_addr_v4)
+ gst_rtsp_address_free (priv->server_addr_v4);
+ if (priv->server_addr_v6)
+ gst_rtsp_address_free (priv->server_addr_v6);
+ if (priv->pool)
+ g_object_unref (priv->pool);
+ gst_object_unref (priv->payloader);
+ gst_object_unref (priv->srcpad);
+ g_free (priv->control);
+ g_mutex_clear (&priv->lock);
+
+ g_hash_table_unref (priv->keys);
+
+ G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_stream_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPStream *stream = GST_RTSP_STREAM (object);
+
+ switch (propid) {
+ case PROP_CONTROL:
+ g_value_take_string (value, gst_rtsp_stream_get_control (stream));
+ break;
+ case PROP_PROFILES:
+ g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
+ break;
+ case PROP_PROTOCOLS:
+ g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_stream_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPStream *stream = GST_RTSP_STREAM (object);
+
+ switch (propid) {
+ case PROP_CONTROL:
+ gst_rtsp_stream_set_control (stream, g_value_get_string (value));
+ break;
+ case PROP_PROFILES:
+ gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
+ break;
+ case PROP_PROTOCOLS:
+ gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+/**
+ * gst_rtsp_stream_new:
+ * @idx: an index
+ * @srcpad: a #GstPad
+ * @payloader: a #GstElement
+ *
+ * Create a new media stream with index @idx that handles RTP data on
+ * @srcpad and has a payloader element @payloader.
+ *
+ * Returns: (transfer full): a new #GstRTSPStream
+ */
+GstRTSPStream *
+gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPStream *stream;
+
+ g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
+ g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
+ g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
+
+ stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
+ priv = stream->priv;
+ priv->idx = idx;
+ priv->payloader = gst_object_ref (payloader);
+ priv->srcpad = gst_object_ref (srcpad);
+
+ return stream;
+}
+
+/**
+ * gst_rtsp_stream_get_index:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the stream index.
+ *
+ * Return: the stream index.
+ */
+guint
+gst_rtsp_stream_get_index (GstRTSPStream * stream)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ return stream->priv->idx;
+}
+
+/**
+ * gst_rtsp_stream_get_pt:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the stream payload type.
+ *
+ * Return: the stream payload type.
+ */
+guint
+gst_rtsp_stream_get_pt (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint pt;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
+
+ return pt;
+}
+
+/**
+ * gst_rtsp_stream_get_srcpad:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the srcpad associated with @stream.
+ *
+ * Returns: (transfer full): the srcpad. Unref after usage.
+ */
+GstPad *
+gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ return gst_object_ref (stream->priv->srcpad);
+}
+
+/**
+ * gst_rtsp_stream_get_control:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the control string to identify this stream.
+ *
+ * Returns: (transfer full): the control string. g_free() after usage.
+ */
+gchar *
+gst_rtsp_stream_get_control (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = g_strdup (priv->control)) == NULL)
+ result = g_strdup_printf ("stream=%u", priv->idx);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_set_control:
+ * @stream: a #GstRTSPStream
+ * @control: a control string
+ *
+ * Set the control string in @stream.
+ */
+void
+gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_free (priv->control);
+ priv->control = g_strdup (control);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_has_control:
+ * @stream: a #GstRTSPStream
+ * @control: a control string
+ *
+ * Check if @stream has the control string @control.
+ *
+ * Returns: %TRUE is @stream has @control as the control string
+ */
+gboolean
+gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->control)
+ res = (g_strcmp0 (priv->control, control) == 0);
+ else {
+ guint streamid;
+
+ if (sscanf (control, "stream=%u", &streamid) > 0)
+ res = (streamid == priv->idx);
+ else
+ res = FALSE;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_mtu:
+ * @stream: a #GstRTSPStream
+ * @mtu: a new MTU
+ *
+ * Configure the mtu in the payloader of @stream to @mtu.
+ */
+void
+gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ GST_LOG_OBJECT (stream, "set MTU %u", mtu);
+
+ g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
+}
+
+/**
+ * gst_rtsp_stream_get_mtu:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured MTU in the payloader of @stream.
+ *
+ * Returns: the MTU of the payloader.
+ */
+guint
+gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint mtu;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
+
+ return mtu;
+}
+
+/* Update the dscp qos property on the udp sinks */
+static void
+update_dscp_qos (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ if (priv->udpsink[0]) {
+ g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
+ NULL);
+ }
+
+ if (priv->udpsink[1]) {
+ g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
+ NULL);
+ }
+}
+
+/**
+ * gst_rtsp_stream_set_dscp_qos:
+ * @stream: a #GstRTSPStream
+ * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
+ *
+ * Configure the dscp qos of the outgoing sockets to @dscp_qos.
+ */
+void
+gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
+
+ if (dscp_qos < -1 || dscp_qos > 63) {
+ GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
+ return;
+ }
+
+ priv->dscp_qos = dscp_qos;
+
+ update_dscp_qos (stream);
+}
+
+/**
+ * gst_rtsp_stream_get_dscp_qos:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured DSCP QoS in of the outgoing sockets.
+ *
+ * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
+ */
+gint
+gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ priv = stream->priv;
+
+ return priv->dscp_qos;
+}
+
+/**
+ * gst_rtsp_stream_is_transport_supported:
+ * @stream: a #GstRTSPStream
+ * @transport: (transfer none): a #GstRTSPTransport
+ *
+ * Check if @transport can be handled by stream
+ *
+ * Returns: %TRUE if @transport can be handled by @stream.
+ */
+gboolean
+gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
+ GstRTSPTransport * transport)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (transport->trans != GST_RTSP_TRANS_RTP)
+ goto unsupported_transmode;
+
+ if (!(transport->profile & priv->profiles))
+ goto unsupported_profile;
+
+ if (!(transport->lower_transport & priv->protocols))
+ goto unsupported_ltrans;
+
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+ /* ERRORS */
+unsupported_transmode:
+ {
+ GST_DEBUG ("unsupported transport mode %d", transport->trans);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+unsupported_profile:
+ {
+ GST_DEBUG ("unsupported profile %d", transport->profile);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+unsupported_ltrans:
+ {
+ GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_set_profiles:
+ * @stream: a #GstRTSPStream
+ * @profiles: the new profiles
+ *
+ * Configure the allowed profiles for @stream.
+ */
+void
+gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->profiles = profiles;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_profiles:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed profiles of @stream.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->profiles;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_protocols:
+ * @stream: a #GstRTSPStream
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @stream.
+ */
+void
+gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
+ GstRTSPLowerTrans protocols)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->protocols = protocols;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_protocols:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed protocols of @stream.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPLowerTrans res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
+ GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->protocols;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_address_pool:
+ * @stream: a #GstRTSPStream
+ * @pool: (transfer none): a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @stream.
+ */
+void
+gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
+ GstRTSPAddressPool * pool)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddressPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ GST_LOG_OBJECT (stream, "set address pool %p", pool);
+
+ g_mutex_lock (&priv->lock);
+ if ((old = priv->pool) != pool)
+ priv->pool = pool ? g_object_ref (pool) : NULL;
+ else
+ old = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_stream_get_address_pool:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @stream.
+ *
+ * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
+ * usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddressPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->pool))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_get_multicast_address:
+ * @stream: a #GstRTSPStream
+ * @family: the #GSocketFamily
+ *
+ * Get the multicast address of @stream for @family.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
+ * or %NULL when no address could be allocated. gst_rtsp_address_free()
+ * after usage.
+ */
+GstRTSPAddress *
+gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
+ GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddress *result;
+ GstRTSPAddress **addrp;
+ GstRTSPAddressFlags flags;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ if (family == G_SOCKET_FAMILY_IPV6) {
+ flags = GST_RTSP_ADDRESS_FLAG_IPV6;
+ addrp = &priv->addr_v6;
+ } else {
+ flags = GST_RTSP_ADDRESS_FLAG_IPV4;
+ addrp = &priv->addr_v4;
+ }
+
+ g_mutex_lock (&priv->lock);
+ if (*addrp == NULL) {
+ if (priv->pool == NULL)
+ goto no_pool;
+
+ flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
+
+ *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
+ if (*addrp == NULL)
+ goto no_address;
+ }
+ result = gst_rtsp_address_copy (*addrp);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+
+ /* ERRORS */
+no_pool:
+ {
+ GST_ERROR_OBJECT (stream, "no address pool specified");
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+no_address:
+ {
+ GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_stream_reserve_address:
+ * @stream: a #GstRTSPStream
+ * @address: an address
+ * @port: a port
+ * @n_ports: n_ports
+ * @ttl: a TTL
+ *
+ * Reserve @address and @port as the address and port of @stream.
+ *
+ * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
+ * the address could be reserved. gst_rtsp_address_free() after usage.
+ */
+GstRTSPAddress *
+gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
+ const gchar * address, guint port, guint n_ports, guint ttl)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddress *result;
+ GInetAddress *addr;
+ GSocketFamily family;
+ GstRTSPAddress **addrp;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (address != NULL, NULL);
+ g_return_val_if_fail (port > 0, NULL);
+ g_return_val_if_fail (n_ports > 0, NULL);
+ g_return_val_if_fail (ttl > 0, NULL);
+
+ priv = stream->priv;
+
+ addr = g_inet_address_new_from_string (address);
+ if (!addr) {
+ GST_ERROR ("failed to get inet addr from %s", address);
+ family = G_SOCKET_FAMILY_IPV4;
+ } else {
+ family = g_inet_address_get_family (addr);
+ g_object_unref (addr);
+ }
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ addrp = &priv->addr_v6;
+ else
+ addrp = &priv->addr_v4;
+
+ g_mutex_lock (&priv->lock);
+ if (*addrp == NULL) {
+ GstRTSPAddressPoolResult res;
+
+ if (priv->pool == NULL)
+ goto no_pool;
+
+ res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
+ port, n_ports, ttl, addrp);
+ if (res != GST_RTSP_ADDRESS_POOL_OK)
+ goto no_address;
+ } else {
+ if (strcmp ((*addrp)->address, address) ||
+ (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
+ (*addrp)->ttl != ttl)
+ goto different_address;
+ }
+ result = gst_rtsp_address_copy (*addrp);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+
+ /* ERRORS */
+no_pool:
+ {
+ GST_ERROR_OBJECT (stream, "no address pool specified");
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+no_address:
+ {
+ GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
+ address);
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+different_address:
+ {
+ GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
+ " reserved", address);
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+}
+
+static gboolean
+alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
+ GSocketFamily family, GstElement * udpsrc_out[2],
+ GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
+ GstRTSPAddress ** server_addr_out)
+{
+ GstStateChangeReturn ret;
+ GstElement *udpsrc0, *udpsrc1;
+ GstElement *udpsink0, *udpsink1;
+ GSocket *rtp_socket = NULL;
+ GSocket *rtcp_socket;
+ gint tmp_rtp, tmp_rtcp;
+ guint count;
+ gint rtpport, rtcpport;
+ GList *rejected_addresses = NULL;
+ GstRTSPAddress *addr = NULL;
+ GInetAddress *inetaddr = NULL;
+ GSocketAddress *rtp_sockaddr = NULL;
+ GSocketAddress *rtcp_sockaddr = NULL;
+ const gchar *multisink_socket;
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ multisink_socket = "socket-v6";
+ else
+ multisink_socket = "socket";
+
+ udpsrc0 = NULL;
+ udpsrc1 = NULL;
+ udpsink0 = NULL;
+ udpsink1 = NULL;
+ count = 0;
+
+ /* Start with random port */
+ tmp_rtp = 0;
+
+ rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ if (!rtcp_socket)
+ goto no_udp_protocol;
+
+ if (*server_addr_out)
+ gst_rtsp_address_free (*server_addr_out);
+
+ /* try to allocate 2 UDP ports, the RTP port should be an even
+ * number and the RTCP port should be the next (uneven) port */
+again:
+
+ if (rtp_socket == NULL) {
+ rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ if (!rtp_socket)
+ goto no_udp_protocol;
+ }
+
+ if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
+ GstRTSPAddressFlags flags;
+
+ if (addr)
+ rejected_addresses = g_list_prepend (rejected_addresses, addr);
+
+ flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
+ if (family == G_SOCKET_FAMILY_IPV6)
+ flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
+ else
+ flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
+
+ addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
+
+ if (addr == NULL)
+ goto no_ports;
+
+ tmp_rtp = addr->port;
+
+ g_clear_object (&inetaddr);
+ inetaddr = g_inet_address_new_from_string (addr->address);
+ } else {
+ if (tmp_rtp != 0) {
+ tmp_rtp += 2;
+ if (++count > 20)
+ goto no_ports;
+ }
+
+ if (inetaddr == NULL)
+ inetaddr = g_inet_address_new_any (family);
+ }
+
+ /* FIXME-WFD : Force to set 19000 as port number */
+ tmp_rtp = 19000;
+
+ rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
+ if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
+ g_object_unref (rtp_sockaddr);
+ goto again;
+ }
+ g_object_unref (rtp_sockaddr);
+
+ rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
+ if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
+ g_clear_object (&rtp_sockaddr);
+ goto socket_error;
+ }
+
+ tmp_rtp =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
+ g_object_unref (rtp_sockaddr);
+
+ /* check if port is even */
+ if ((tmp_rtp & 1) != 0) {
+ /* port not even, close and allocate another */
+ tmp_rtp++;
+ g_clear_object (&rtp_socket);
+ goto again;
+ }
+
+ /* set port */
+ tmp_rtcp = tmp_rtp + 1;
+
+ rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
+ if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
+ g_object_unref (rtcp_sockaddr);
+ g_clear_object (&rtp_socket);
+ goto again;
+ }
+ g_object_unref (rtcp_sockaddr);
+
+ g_clear_object (&inetaddr);
+
+ udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
+ udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
+
+ if (udpsrc0 == NULL || udpsrc1 == NULL)
+ goto no_udp_protocol;
+
+ g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
+ g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
+
+ ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto element_error;
+ ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto element_error;
+
+ /* all fine, do port check */
+ g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
+ g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
+
+ /* this should not happen... */
+ if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
+ goto port_error;
+
+ if (udpsink_out[0])
+ udpsink0 = udpsink_out[0];
+ else
+ udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
+
+ if (!udpsink0)
+ goto no_udp_protocol;
+
+ g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
+
+ if (udpsink_out[1])
+ udpsink1 = udpsink_out[1];
+ else
+ udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
+
+ if (!udpsink1)
+ goto no_udp_protocol;
+
+ g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
+
+ g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
+ g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
+
+ /* we keep these elements, we will further configure them when the
+ * client told us to really use the UDP ports. */
+ udpsrc_out[0] = udpsrc0;
+ udpsrc_out[1] = udpsrc1;
+ udpsink_out[0] = udpsink0;
+ udpsink_out[1] = udpsink1;
+ server_port_out->min = rtpport;
+ server_port_out->max = rtcpport;
+
+ *server_addr_out = addr;
+ g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
+
+ g_object_unref (rtp_socket);
+ g_object_unref (rtcp_socket);
+
+ return TRUE;
+
+ /* ERRORS */
+no_udp_protocol:
+ {
+ goto cleanup;
+ }
+no_ports:
+ {
+ goto cleanup;
+ }
+port_error:
+ {
+ goto cleanup;
+ }
+socket_error:
+ {
+ goto cleanup;
+ }
+element_error:
+ {
+ goto cleanup;
+ }
+cleanup:
+ {
+ if (udpsrc0) {
+ gst_element_set_state (udpsrc0, GST_STATE_NULL);
+ gst_object_unref (udpsrc0);
+ }
+ if (udpsrc1) {
+ gst_element_set_state (udpsrc1, GST_STATE_NULL);
+ gst_object_unref (udpsrc1);
+ }
+ if (udpsink0) {
+ gst_element_set_state (udpsink0, GST_STATE_NULL);
+ gst_object_unref (udpsink0);
+ }
+ if (inetaddr)
+ g_object_unref (inetaddr);
+ g_list_free_full (rejected_addresses,
+ (GDestroyNotify) gst_rtsp_address_free);
+ if (addr)
+ gst_rtsp_address_free (addr);
+ if (rtp_socket)
+ g_object_unref (rtp_socket);
+ if (rtcp_socket)
+ g_object_unref (rtcp_socket);
+ return FALSE;
+ }
+}
+
+/* must be called with lock */
+static gboolean
+alloc_ports (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
+ G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
+ &priv->server_port_v4, &priv->server_addr_v4);
+
+ /* FIXME-WFD : force to disable ipv6 mode in WFD mode */
+#if 0
+ priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
+ G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
+ &priv->server_port_v6, &priv->server_addr_v6);
+#else
+ priv->have_ipv6 = FALSE;
+#endif
+
+ return priv->have_ipv4 || priv->have_ipv6;
+}
+
+/**
+ * gst_rtsp_stream_get_server_port:
+ * @stream: a #GstRTSPStream
+ * @server_port: (out): result server port
+ * @family: the port family to get
+ *
+ * Fill @server_port with the port pair used by the server. This function can
+ * only be called when @stream has been joined.
+ */
+void
+gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
+ GstRTSPRange * server_port, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ priv = stream->priv;
+ g_return_if_fail (priv->is_joined);
+
+ g_mutex_lock (&priv->lock);
+ if (family == G_SOCKET_FAMILY_IPV4) {
+ if (server_port)
+ *server_port = priv->server_port_v4;
+ } else {
+ if (server_port)
+ *server_port = priv->server_port_v6;
+ }
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_rtpsession:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the RTP session of this stream.
+ *
+ * Returns: (transfer full): The RTP session of this stream. Unref after usage.
+ */
+GObject *
+gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GObject *session;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((session = priv->session))
+ g_object_ref (session);
+ g_mutex_unlock (&priv->lock);
+
+ return session;
+}
+
+/**
+ * gst_rtsp_stream_get_ssrc:
+ * @stream: a #GstRTSPStream
+ * @ssrc: (out): result ssrc
+ *
+ * Get the SSRC used by the RTP session of this stream. This function can only
+ * be called when @stream has been joined.
+ */
+void
+gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ priv = stream->priv;
+ g_return_if_fail (priv->is_joined);
+
+ g_mutex_lock (&priv->lock);
+ if (ssrc && priv->session)
+ g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
+ g_mutex_unlock (&priv->lock);
+}
+
+/* executed from streaming thread */
+static void
+caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstCaps *newcaps, *oldcaps;
+
+ newcaps = gst_pad_get_current_caps (pad);
+
+ GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
+ newcaps);
+
+ g_mutex_lock (&priv->lock);
+ oldcaps = priv->caps;
+ priv->caps = newcaps;
+ g_mutex_unlock (&priv->lock);
+
+ if (oldcaps)
+ gst_caps_unref (oldcaps);
+}
+
+static void
+dump_structure (const GstStructure * s)
+{
+ gchar *sstr;
+
+ sstr = gst_structure_to_string (s);
+ GST_INFO ("structure: %s", sstr);
+ g_free (sstr);
+}
+
+static GstRTSPStreamTransport *
+find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GList *walk;
+ GstRTSPStreamTransport *result = NULL;
+ const gchar *tmp;
+ gchar *dest;
+ guint port;
+
+ if (rtcp_from == NULL)
+ return NULL;
+
+ tmp = g_strrstr (rtcp_from, ":");
+ if (tmp == NULL)
+ return NULL;
+
+ port = atoi (tmp + 1);
+ dest = g_strndup (rtcp_from, tmp - rtcp_from);
+
+ g_mutex_lock (&priv->lock);
+ GST_INFO ("finding %s:%d in %d transports", dest, port,
+ g_list_length (priv->transports));
+
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *trans = walk->data;
+ const GstRTSPTransport *tr;
+ gint min, max;
+
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+
+ min = tr->client_port.min;
+ max = tr->client_port.max;
+
+ if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
+ result = trans;
+ break;
+ }
+ }
+ if (result)
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ g_free (dest);
+
+ return result;
+}
+
+static GstRTSPStreamTransport *
+check_transport (GObject * source, GstRTSPStream * stream)
+{
+ GstStructure *stats;
+ GstRTSPStreamTransport *trans;
+
+ /* see if we have a stream to match with the origin of the RTCP packet */
+ trans = g_object_get_qdata (source, ssrc_stream_map_key);
+ if (trans == NULL) {
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ const gchar *rtcp_from;
+
+ dump_structure (stats);
+
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_RTCP_STATS], 0, stats);
+
+ rtcp_from = gst_structure_get_string (stats, "rtcp-from");
+ if ((trans = find_transport (stream, rtcp_from))) {
+ GST_INFO ("%p: found transport %p for source %p", stream, trans,
+ source);
+ g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
+ g_object_unref);
+ }
+ gst_structure_free (stats);
+ }
+ }
+ return trans;
+}
+
+
+static void
+on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ GST_INFO ("%p: new source %p", stream, source);
+
+ trans = check_transport (source, stream);
+
+ if (trans)
+ GST_INFO ("%p: source %p for transport %p", stream, source, trans);
+}
+
+static void
+on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GST_INFO ("%p: new SDES %p", stream, source);
+}
+
+static void
+on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ trans = check_transport (source, stream);
+
+ if (trans) {
+ GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
+ gst_rtsp_stream_transport_keep_alive (trans);
+ }
+ {
+ GstStructure *stats;
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_RTCP_STATS], 0, stats);
+
+#ifdef DUMP_STATS
+ dump_structure (stats);
+#endif
+ gst_structure_free (stats);
+ }
+ }
+}
+
+static void
+on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GST_INFO ("%p: source %p bye", stream, source);
+}
+
+static void
+on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ GST_INFO ("%p: source %p bye timeout", stream, source);
+
+ if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
+ gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+ g_object_set_qdata (source, ssrc_stream_map_key, NULL);
+ }
+}
+
+static void
+on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ GST_INFO ("%p: source %p timeout", stream, source);
+
+ if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
+ gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+ g_object_set_qdata (source, ssrc_stream_map_key, NULL);
+ }
+}
+
+static void
+clear_tr_cache (GstRTSPStreamPrivate * priv)
+{
+ g_list_foreach (priv->tr_cache, (GFunc) g_object_unref, NULL);
+ g_list_free (priv->tr_cache);
+ priv->tr_cache = NULL;
+}
+
+static GstFlowReturn
+handle_new_sample (GstAppSink * sink, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GList *walk;
+ GstSample *sample;
+ GstBuffer *buffer;
+ GstRTSPStream *stream;
+ gboolean is_rtp;
+
+ sample = gst_app_sink_pull_sample (sink);
+ if (!sample)
+ return GST_FLOW_OK;
+
+ stream = (GstRTSPStream *) user_data;
+ priv = stream->priv;
+ buffer = gst_sample_get_buffer (sample);
+
+ is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
+
+ g_mutex_lock (&priv->lock);
+ if (priv->tr_cache_cookie != priv->transports_cookie) {
+ clear_tr_cache (priv);
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+ priv->tr_cache = g_list_prepend (priv->tr_cache, g_object_ref (tr));
+ }
+ priv->tr_cache_cookie = priv->transports_cookie;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ for (walk = priv->tr_cache; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+
+ if (is_rtp) {
+ gst_rtsp_stream_transport_send_rtp (tr, buffer);
+ } else {
+ gst_rtsp_stream_transport_send_rtcp (tr, buffer);
+ }
+ }
+ gst_sample_unref (sample);
+
+ return GST_FLOW_OK;
+}
+
+static GstAppSinkCallbacks sink_cb = {
+ NULL, /* not interested in EOS */
+ NULL, /* not interested in preroll samples */
+ handle_new_sample,
+};
+
+static GstElement *
+get_rtp_encoder (GstRTSPStream * stream, guint session)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ if (priv->srtpenc == NULL) {
+ gchar *name;
+
+ name = g_strdup_printf ("srtpenc_%u", session);
+ priv->srtpenc = gst_element_factory_make ("srtpenc", name);
+ g_free (name);
+
+ g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
+ }
+ return gst_object_ref (priv->srtpenc);
+}
+
+static GstElement *
+request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstElement *oldenc, *enc;
+ GstPad *pad;
+ gchar *name;
+
+ if (priv->idx != session)
+ return NULL;
+
+ GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
+
+ oldenc = priv->srtpenc;
+ enc = get_rtp_encoder (stream, session);
+ name = g_strdup_printf ("rtp_sink_%d", session);
+ pad = gst_element_get_request_pad (enc, name);
+ g_free (name);
+ gst_object_unref (pad);
+
+ if (oldenc == NULL)
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
+ enc);
+
+ return enc;
+}
+
+static GstElement *
+request_rtcp_encoder (GstElement * rtpbin, guint session,
+ GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstElement *oldenc, *enc;
+ GstPad *pad;
+ gchar *name;
+
+ if (priv->idx != session)
+ return NULL;
+
+ GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
+
+ oldenc = priv->srtpenc;
+ enc = get_rtp_encoder (stream, session);
+ name = g_strdup_printf ("rtcp_sink_%d", session);
+ pad = gst_element_get_request_pad (enc, name);
+ g_free (name);
+ gst_object_unref (pad);
+
+ if (oldenc == NULL)
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
+ enc);
+
+ return enc;
+}
+
+static GstCaps *
+request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstCaps *caps;
+
+ GST_DEBUG ("request key %08x", ssrc);
+
+ g_mutex_lock (&priv->lock);
+ if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
+ gst_caps_ref (caps);
+ g_mutex_unlock (&priv->lock);
+
+ return caps;
+}
+
+static GstElement *
+request_rtcp_decoder (GstElement * rtpbin, guint session,
+ GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ if (priv->idx != session)
+ return NULL;
+
+ if (priv->srtpdec == NULL) {
+ gchar *name;
+
+ name = g_strdup_printf ("srtpdec_%u", session);
+ priv->srtpdec = gst_element_factory_make ("srtpdec", name);
+ g_free (name);
+
+ g_signal_connect (priv->srtpdec, "request-key",
+ (GCallback) request_key, stream);
+ }
+ return gst_object_ref (priv->srtpdec);
+}
+
+/**
+ * gst_rtsp_stream_join_bin:
+ * @stream: a #GstRTSPStream
+ * @bin: (transfer none): a #GstBin to join
+ * @rtpbin: (transfer none): a rtpbin element in @bin
+ * @state: the target state of the new elements
+ *
+ * Join the #GstBin @bin that contains the element @rtpbin.
+ *
+ * @stream will link to @rtpbin, which must be inside @bin. The elements
+ * added to @bin will be set to the state given in @state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
+ GstElement * rtpbin, GstState state)
+{
+ GstRTSPStreamPrivate *priv;
+ gint i;
+ guint idx;
+ gchar *name;
+ GstPad *pad, *sinkpad, *selpad;
+ GstPadLinkReturn ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
+ g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->is_joined)
+ goto was_joined;
+
+ /* create a session with the same index as the stream */
+ idx = priv->idx;
+
+ GST_INFO ("stream %p joining bin as session %u", stream, idx);
+
+ if (!alloc_ports (stream))
+ goto no_ports;
+
+ /* update the dscp qos field in the sinks */
+ update_dscp_qos (stream);
+
+ if (priv->profiles & GST_RTSP_PROFILE_SAVP
+ || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
+ /* For SRTP */
+ g_signal_connect (rtpbin, "request-rtp-encoder",
+ (GCallback) request_rtp_encoder, stream);
+ g_signal_connect (rtpbin, "request-rtcp-encoder",
+ (GCallback) request_rtcp_encoder, stream);
+ g_signal_connect (rtpbin, "request-rtcp-decoder",
+ (GCallback) request_rtcp_decoder, stream);
+ }
+
+ /* get a pad for sending RTP */
+ name = g_strdup_printf ("send_rtp_sink_%u", idx);
+ priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
+ g_free (name);
+ /* link the RTP pad to the session manager, it should not really fail unless
+ * this is not really an RTP pad */
+ ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
+ if (ret != GST_PAD_LINK_OK)
+ goto link_failed;
+
+ /* get pads from the RTP session element for sending and receiving
+ * RTP/RTCP*/
+ name = g_strdup_printf ("send_rtp_src_%u", idx);
+ priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
+ g_free (name);
+ name = g_strdup_printf ("send_rtcp_src_%u", idx);
+ priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
+ g_free (name);
+ name = g_strdup_printf ("recv_rtp_sink_%u", idx);
+ priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
+ g_free (name);
+ name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
+ priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
+ g_free (name);
+
+ /* get the session */
+ g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
+
+ g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
+ stream);
+ g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
+ stream);
+ g_signal_connect (priv->session, "on-ssrc-active",
+ (GCallback) on_ssrc_active, stream);
+ g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
+ stream);
+ g_signal_connect (priv->session, "on-bye-timeout",
+ (GCallback) on_bye_timeout, stream);
+ g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
+ stream);
+
+ for (i = 0; i < 2; i++) {
+ GstPad *teepad, *queuepad;
+ /* For the sender we create this bit of pipeline for both
+ * RTP and RTCP. Sync and preroll are enabled on udpsink so
+ * we need to add a queue before appsink to make the pipeline
+ * not block. For the TCP case, we want to pump data to the
+ * client as fast as possible anyway.
+ *
+ * .--------. .-----. .---------.
+ * | rtpbin | | tee | | udpsink |
+ * | send->sink src->sink |
+ * '--------' | | '---------'
+ * | | .---------. .---------.
+ * | | | queue | | appsink |
+ * | src->sink src->sink |
+ * '-----' '---------' '---------'
+ *
+ * When only UDP is allowed, we skip the tee, queue and appsink and link the
+ * udpsink directly to the session.
+ */
+ /* add udpsink */
+ gst_bin_add (bin, priv->udpsink[i]);
+ sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
+
+ if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
+ /* make tee for RTP/RTCP */
+ priv->tee[i] = gst_element_factory_make ("tee", NULL);
+ gst_bin_add (bin, priv->tee[i]);
+
+ /* and link to rtpbin send pad */
+ pad = gst_element_get_static_pad (priv->tee[i], "sink");
+ gst_pad_link (priv->send_src[i], pad);
+ gst_object_unref (pad);
+
+ /* link tee to udpsink */
+ teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
+ gst_pad_link (teepad, sinkpad);
+ gst_object_unref (teepad);
+
+ /* make queue */
+ priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
+ gst_bin_add (bin, priv->appqueue[i]);
+ /* and link to tee */
+ teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
+ pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
+ gst_pad_link (teepad, pad);
+ gst_object_unref (pad);
+ gst_object_unref (teepad);
+
+ /* make appsink */
+ priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
+ gst_bin_add (bin, priv->appsink[i]);
+ gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
+ &sink_cb, stream, NULL);
+ /* and link to queue */
+ queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
+ pad = gst_element_get_static_pad (priv->appsink[i], "sink");
+ gst_pad_link (queuepad, pad);
+ gst_object_unref (pad);
+ gst_object_unref (queuepad);
+ } else {
+ /* else only udpsink needed, link it to the session */
+ gst_pad_link (priv->send_src[i], sinkpad);
+ }
+ gst_object_unref (sinkpad);
+
+ /* For the receiver we create this bit of pipeline for both
+ * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
+ * and it is all funneled into the rtpbin receive pad.
+ *
+ * .--------. .--------. .--------.
+ * | udpsrc | | funnel | | rtpbin |
+ * | src->sink src->sink |
+ * '--------' | | '--------'
+ * .--------. | |
+ * | appsrc | | |
+ * | src->sink |
+ * '--------' '--------'
+ */
+ /* make funnel for the RTP/RTCP receivers */
+ priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
+ gst_bin_add (bin, priv->funnel[i]);
+
+ pad = gst_element_get_static_pad (priv->funnel[i], "src");
+ gst_pad_link (pad, priv->recv_sink[i]);
+ gst_object_unref (pad);
+
+ if (priv->udpsrc_v4[i]) {
+ /* we set and keep these to playing so that they don't cause NO_PREROLL return
+ * values */
+ gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
+ gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
+ /* add udpsrc */
+ gst_bin_add (bin, priv->udpsrc_v4[i]);
+
+ /* and link to the funnel v4 */
+ selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+ }
+
+ if (priv->udpsrc_v6[i]) {
+ gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
+ gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
+ gst_bin_add (bin, priv->udpsrc_v6[i]);
+
+ /* and link to the funnel v6 */
+ selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+ }
+
+ if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
+ /* make and add appsrc */
+ priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
+ gst_bin_add (bin, priv->appsrc[i]);
+ /* and link to the funnel */
+ selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (priv->appsrc[i], "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+ }
+
+ /* check if we need to set to a special state */
+ if (state != GST_STATE_NULL) {
+ if (priv->udpsink[i])
+ gst_element_set_state (priv->udpsink[i], state);
+ if (priv->appsink[i])
+ gst_element_set_state (priv->appsink[i], state);
+ if (priv->appqueue[i])
+ gst_element_set_state (priv->appqueue[i], state);
+ if (priv->tee[i])
+ gst_element_set_state (priv->tee[i], state);
+ if (priv->funnel[i])
+ gst_element_set_state (priv->funnel[i], state);
+ if (priv->appsrc[i])
+ gst_element_set_state (priv->appsrc[i], state);
+ }
+ }
+
+ /* be notified of caps changes */
+ priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
+ (GCallback) caps_notify, stream);
+
+ priv->is_joined = TRUE;
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+ /* ERRORS */
+was_joined:
+ {
+ g_mutex_unlock (&priv->lock);
+ return TRUE;
+ }
+no_ports:
+ {
+ g_mutex_unlock (&priv->lock);
+ GST_WARNING ("failed to allocate ports %u", idx);
+ return FALSE;
+ }
+link_failed:
+ {
+ GST_WARNING ("failed to link stream %u", idx);
+ gst_object_unref (priv->send_rtp_sink);
+ priv->send_rtp_sink = NULL;
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_leave_bin:
+ * @stream: a #GstRTSPStream
+ * @bin: (transfer none): a #GstBin
+ * @rtpbin: (transfer none): a rtpbin #GstElement
+ *
+ * Remove the elements of @stream from @bin.
+ *
+ * Return: %TRUE on success.
+ */
+gboolean
+gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
+ GstElement * rtpbin)
+{
+ GstRTSPStreamPrivate *priv;
+ gint i;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
+ g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (!priv->is_joined)
+ goto was_not_joined;
+
+ /* all transports must be removed by now */
+ g_return_val_if_fail (priv->transports == NULL, FALSE);
+
+ clear_tr_cache (priv);
+
+ GST_INFO ("stream %p leaving bin", stream);
+
+ gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
+ g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
+ gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
+ gst_object_unref (priv->send_rtp_sink);
+ priv->send_rtp_sink = NULL;
+
+ for (i = 0; i < 2; i++) {
+ if (priv->udpsink[i])
+ gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
+ if (priv->appsink[i])
+ gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
+ if (priv->appqueue[i])
+ gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
+ if (priv->tee[i])
+ gst_element_set_state (priv->tee[i], GST_STATE_NULL);
+ if (priv->funnel[i])
+ gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
+ if (priv->appsrc[i])
+ gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
+ if (priv->udpsrc_v4[i]) {
+ /* and set udpsrc to NULL now before removing */
+ gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
+ gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
+ /* removing them should also nicely release the request
+ * pads when they finalize */
+ gst_bin_remove (bin, priv->udpsrc_v4[i]);
+ }
+ if (priv->udpsrc_v6[i]) {
+ gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
+ gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
+ gst_bin_remove (bin, priv->udpsrc_v6[i]);
+ }
+ if (priv->udpsink[i])
+ gst_bin_remove (bin, priv->udpsink[i]);
+ if (priv->appsrc[i])
+ gst_bin_remove (bin, priv->appsrc[i]);
+ if (priv->appsink[i])
+ gst_bin_remove (bin, priv->appsink[i]);
+ if (priv->appqueue[i])
+ gst_bin_remove (bin, priv->appqueue[i]);
+ if (priv->tee[i])
+ gst_bin_remove (bin, priv->tee[i]);
+ if (priv->funnel[i])
+ gst_bin_remove (bin, priv->funnel[i]);
+
+ gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
+ gst_object_unref (priv->recv_sink[i]);
+ priv->recv_sink[i] = NULL;
+
+ priv->udpsrc_v4[i] = NULL;
+ priv->udpsrc_v6[i] = NULL;
+ priv->udpsink[i] = NULL;
+ priv->appsrc[i] = NULL;
+ priv->appsink[i] = NULL;
+ priv->appqueue[i] = NULL;
+ priv->tee[i] = NULL;
+ priv->funnel[i] = NULL;
+ }
+ gst_object_unref (priv->send_src[0]);
+ priv->send_src[0] = NULL;
+
+ gst_element_release_request_pad (rtpbin, priv->send_src[1]);
+ gst_object_unref (priv->send_src[1]);
+ priv->send_src[1] = NULL;
+
+ g_object_unref (priv->session);
+ priv->session = NULL;
+ if (priv->caps)
+ gst_caps_unref (priv->caps);
+ priv->caps = NULL;
+
+ if (priv->srtpenc)
+ gst_object_unref (priv->srtpenc);
+ if (priv->srtpdec)
+ gst_object_unref (priv->srtpdec);
+
+ priv->is_joined = FALSE;
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+was_not_joined:
+ {
+ g_mutex_unlock (&priv->lock);
+ return TRUE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_get_rtpinfo:
+ * @stream: a #GstRTSPStream
+ * @rtptime: (allow-none): result RTP timestamp
+ * @seq: (allow-none): result RTP seqnum
+ * @clock_rate: (allow-none): the clock rate
+ * @running_time: (allow-none): result running-time
+ *
+ * Retrieve the current rtptime, seq and running-time. This is used to
+ * construct a RTPInfo reply header.
+ *
+ * Returns: %TRUE when rtptime, seq and running-time could be determined.
+ */
+gboolean
+gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
+ guint * rtptime, guint * seq, guint * clock_rate,
+ GstClockTime * running_time)
+{
+ GstRTSPStreamPrivate *priv;
+ GstStructure *stats;
+ GObjectClass *payobjclass;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
+
+ g_mutex_lock (&priv->lock);
+
+ if (g_object_class_find_property (payobjclass, "stats")) {
+ g_object_get (priv->payloader, "stats", &stats, NULL);
+ if (stats == NULL)
+ goto no_stats;
+
+ if (seq)
+ gst_structure_get_uint (stats, "seqnum", seq);
+
+ if (rtptime)
+ gst_structure_get_uint (stats, "timestamp", rtptime);
+
+ if (running_time)
+ gst_structure_get_clock_time (stats, "running-time", running_time);
+
+ if (clock_rate) {
+ gst_structure_get_uint (stats, "clock-rate", clock_rate);
+ if (*clock_rate == 0 && running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ gst_structure_free (stats);
+ } else {
+ if (!g_object_class_find_property (payobjclass, "seqnum") ||
+ !g_object_class_find_property (payobjclass, "timestamp"))
+ goto no_stats;
+
+ if (seq)
+ g_object_get (priv->payloader, "seqnum", seq, NULL);
+
+ if (rtptime)
+ g_object_get (priv->payloader, "timestamp", rtptime, NULL);
+
+ if (running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+ /* ERRORS */
+no_stats:
+ {
+ GST_WARNING ("Could not get payloader stats");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_get_caps:
+ * @stream: a #GstRTSPStream
+ *
+ * Retrieve the current caps of @stream.
+ *
+ * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
+ * after usage.
+ */
+GstCaps *
+gst_rtsp_stream_get_caps (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstCaps *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->caps))
+ gst_caps_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_recv_rtp:
+ * @stream: a #GstRTSPStream
+ * @buffer: (transfer full): a #GstBuffer
+ *
+ * Handle an RTP buffer for the stream. This method is usually called when a
+ * message has been received from a client using the TCP transport.
+ *
+ * This function takes ownership of @buffer.
+ *
+ * Returns: a GstFlowReturn.
+ */
+GstFlowReturn
+gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
+{
+ GstRTSPStreamPrivate *priv;
+ GstFlowReturn ret;
+ GstElement *element;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
+ priv = stream->priv;
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+ g_return_val_if_fail (priv->is_joined, FALSE);
+
+ g_mutex_lock (&priv->lock);
+ if (priv->appsrc[0])
+ element = gst_object_ref (priv->appsrc[0]);
+ else
+ element = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (element) {
+ ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
+ gst_object_unref (element);
+ } else {
+ ret = GST_FLOW_OK;
+ }
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_recv_rtcp:
+ * @stream: a #GstRTSPStream
+ * @buffer: (transfer full): a #GstBuffer
+ *
+ * Handle an RTCP buffer for the stream. This method is usually called when a
+ * message has been received from a client using the TCP transport.
+ *
+ * This function takes ownership of @buffer.
+ *
+ * Returns: a GstFlowReturn.
+ */
+GstFlowReturn
+gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
+{
+ GstRTSPStreamPrivate *priv;
+ GstFlowReturn ret;
+ GstElement *element;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
+ priv = stream->priv;
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+ g_return_val_if_fail (priv->is_joined, FALSE);
+
+ g_mutex_lock (&priv->lock);
+ if (priv->appsrc[1])
+ element = gst_object_ref (priv->appsrc[1]);
+ else
+ element = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (element) {
+ ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
+ gst_object_unref (element);
+ } else {
+ ret = GST_FLOW_OK;
+ gst_buffer_unref (buffer);
+ }
+ return ret;
+}
+
+/* must be called with lock */
+static gboolean
+update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
+ gboolean add)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ const GstRTSPTransport *tr;
+
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+
+ switch (tr->lower_transport) {
+ case GST_RTSP_LOWER_TRANS_UDP:
+ case GST_RTSP_LOWER_TRANS_UDP_MCAST:
+ {
+ gchar *dest;
+ gint min, max;
+ guint ttl = 0;
+
+ dest = tr->destination;
+ if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ min = tr->port.min;
+ max = tr->port.max;
+ ttl = tr->ttl;
+ } else {
+ min = tr->client_port.min;
+ max = tr->client_port.max;
+ }
+
+ if (add) {
+ if (ttl > 0) {
+ GST_INFO ("setting ttl-mc %d", ttl);
+ g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
+ g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
+ }
+ GST_INFO ("adding %s:%d-%d", dest, min, max);
+ g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
+ g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
+ priv->transports = g_list_prepend (priv->transports, trans);
+ } else {
+ GST_INFO ("removing %s:%d-%d", dest, min, max);
+ g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
+ g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
+ priv->transports = g_list_remove (priv->transports, trans);
+ }
+ priv->transports_cookie++;
+ break;
+ }
+ case GST_RTSP_LOWER_TRANS_TCP:
+ if (add) {
+ GST_INFO ("adding TCP %s", tr->destination);
+ priv->transports = g_list_prepend (priv->transports, trans);
+ } else {
+ GST_INFO ("removing TCP %s", tr->destination);
+ priv->transports = g_list_remove (priv->transports, trans);
+ }
+ priv->transports_cookie++;
+ break;
+ default:
+ goto unknown_transport;
+ }
+ return TRUE;
+
+ /* ERRORS */
+unknown_transport:
+ {
+ GST_INFO ("Unknown transport %d", tr->lower_transport);
+ return FALSE;
+ }
+}
+
+
+/**
+ * gst_rtsp_stream_add_transport:
+ * @stream: a #GstRTSPStream
+ * @trans: (transfer none): a #GstRTSPStreamTransport
+ *
+ * Add the transport in @trans to @stream. The media of @stream will
+ * then also be send to the values configured in @trans.
+ *
+ * @stream must be joined to a bin.
+ *
+ * @trans must contain a valid #GstRTSPTransport.
+ *
+ * Returns: %TRUE if @trans was added
+ */
+gboolean
+gst_rtsp_stream_add_transport (GstRTSPStream * stream,
+ GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ priv = stream->priv;
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+ g_return_val_if_fail (priv->is_joined, FALSE);
+
+ g_mutex_lock (&priv->lock);
+ res = update_transport (stream, trans, TRUE);
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_remove_transport:
+ * @stream: a #GstRTSPStream
+ * @trans: (transfer none): a #GstRTSPStreamTransport
+ *
+ * Remove the transport in @trans from @stream. The media of @stream will
+ * not be sent to the values configured in @trans.
+ *
+ * @stream must be joined to a bin.
+ *
+ * @trans must contain a valid #GstRTSPTransport.
+ *
+ * Returns: %TRUE if @trans was removed
+ */
+gboolean
+gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
+ GstRTSPStreamTransport * trans)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ priv = stream->priv;
+ g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+ g_return_val_if_fail (priv->is_joined, FALSE);
+
+ g_mutex_lock (&priv->lock);
+ res = update_transport (stream, trans, FALSE);
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_update_crypto:
+ * @stream: a #GstRTSPStream
+ * @ssrc: the SSRC
+ * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
+ *
+ * Update the new crypto information for @ssrc in @stream. If information
+ * for @ssrc did not exist, it will be added. If information
+ * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
+ * be removed from @stream.
+ *
+ * Returns: %TRUE if @crypto could be updated
+ */
+gboolean
+gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
+ guint ssrc, GstCaps * crypto)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
+
+ priv = stream->priv;
+
+ GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
+
+ g_mutex_lock (&priv->lock);
+ if (crypto)
+ g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
+ gst_caps_ref (crypto));
+ else
+ g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_get_rtp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+ GSocket *socket;
+ const gchar *name;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+ g_return_val_if_fail (priv->udpsink[0], NULL);
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ name = "socket-v6";
+ else
+ name = "socket";
+
+ g_object_get (priv->udpsink[0], name, &socket, NULL);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_get_rtcp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTCP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+ GSocket *socket;
+ const gchar *name;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+ g_return_val_if_fail (priv->udpsink[1], NULL);
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ name = "socket-v6";
+ else
+ name = "socket";
+
+ g_object_get (priv->udpsink[1], name, &socket, NULL);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_set_seqnum:
+ * @stream: a #GstRTSPStream
+ * @seqnum: a new sequence number
+ *
+ * Configure the sequence number in the payloader of @stream to @seqnum.
+ */
+void
+gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
+}
+
+/**
+ * gst_rtsp_stream_get_seqnum:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured sequence number in the payloader of @stream.
+ *
+ * Returns: the sequence number of the payloader.
+ */
+guint16
+gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint seqnum;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
+
+ return seqnum;
+}
+
+guint64
+gst_rtsp_stream_get_udp_sent_bytes (GstRTSPStream *stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint64 bytes = 0;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->udpsink[0]), "bytes-to-serve", &bytes, NULL);
+
+ return bytes;
+}
+
+/**
+ * gst_rtsp_stream_transport_filter:
+ * @stream: a #GstRTSPStream
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each transport managed by @stream. The result value of @func
+ * determines what happens to the transport. @func will be called with @stream
+ * locked so no further actions on @stream can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
+ * @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
+ *
+ * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
+ * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
+ GstRTSPStreamTransportFilterFunc func, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GList *result, *walk, *next;
+ GHashTable *visited;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ g_mutex_lock (&priv->lock);
+restart:
+ cookie = priv->transports_cookie;
+ for (walk = priv->transports; walk; walk = next) {
+ GstRTSPStreamTransport *trans = walk->data;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ next = g_list_next (walk);
+
+ if (func) {
+ /* only visit each transport once */
+ if (g_hash_table_contains (visited, trans))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (trans));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (stream, trans, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->transports_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ update_transport (stream, trans, FALSE);
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (trans));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
+
+static GstPadProbeReturn
+pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPStream *stream;
+
+ stream = user_data;
+ priv = stream->priv;
+
+ GST_DEBUG_OBJECT (pad, "now blocking");
+
+ g_mutex_lock (&priv->lock);
+ priv->blocking = TRUE;
+ g_mutex_unlock (&priv->lock);
+
+ gst_element_post_message (priv->payloader,
+ gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
+ gst_structure_new_empty ("GstRTSPStreamBlocking")));
+
+ return GST_PAD_PROBE_OK;
+}
+
+/**
+ * gst_rtsp_stream_set_blocked:
+ * @stream: a #GstRTSPStream
+ * @blocked: boolean indicating we should block or unblock
+ *
+ * Blocks or unblocks the dataflow on @stream.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (blocked) {
+ priv->blocking = FALSE;
+ if (priv->blocked_id == 0) {
+ priv->blocked_id = gst_pad_add_probe (priv->srcpad,
+ GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
+ GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
+ g_object_ref (stream), g_object_unref);
+ }
+ } else {
+ if (priv->blocked_id != 0) {
+ gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
+ priv->blocked_id = 0;
+ priv->blocking = FALSE;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_is_blocking:
+ * @stream: a #GstRTSPStream
+ *
+ * Check if @stream is blocking on a #GstBuffer.
+ *
+ * Returns: %TRUE if @stream is blocking
+ */
+gboolean
+gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = priv->blocking;
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_query_position:
+ * @stream: a #GstRTSPStream
+ *
+ * Query the position of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the position could be queried
+ */
+gboolean
+gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
+{
+ GstRTSPStreamPrivate *priv;
+ GstElement *sink;
+ gboolean ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((sink = priv->udpsink[0]))
+ gst_object_ref (sink);
+ g_mutex_unlock (&priv->lock);
+
+ if (!sink)
+ return FALSE;
+
+ ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
+ gst_object_unref (sink);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_query_stop:
+ * @stream: a #GstRTSPStream
+ *
+ * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the stop could be queried
+ */
+gboolean
+gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
+{
+ GstRTSPStreamPrivate *priv;
+ GstElement *sink;
+ GstQuery *query;
+ gboolean ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((sink = priv->udpsink[0]))
+ gst_object_ref (sink);
+ g_mutex_unlock (&priv->lock);
+
+ if (!sink)
+ return FALSE;
+
+ query = gst_query_new_segment (GST_FORMAT_TIME);
+ if ((ret = gst_element_query (sink, query))) {
+ GstFormat format;
+
+ gst_query_parse_segment (query, NULL, &format, NULL, stop);
+ if (format != GST_FORMAT_TIME)
+ *stop = -1;
+ }
+ gst_query_unref (query);
+ gst_object_unref (sink);
+
+ return ret;
+
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtsprange.h>
+#include <gst/rtsp/gstrtspurl.h>
+#include <gio/gio.h>
+
+#ifndef __GST_RTSP_STREAM_H__
+#define __GST_RTSP_STREAM_H__
+
+G_BEGIN_DECLS
+
+/* types for the media stream */
+#define GST_TYPE_RTSP_STREAM (gst_rtsp_stream_get_type ())
+#define GST_IS_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM))
+#define GST_IS_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM))
+#define GST_RTSP_STREAM_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
+#define GST_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStream))
+#define GST_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
+#define GST_RTSP_STREAM_CAST(obj) ((GstRTSPStream*)(obj))
+#define GST_RTSP_STREAM_CLASS_CAST(klass) ((GstRTSPStreamClass*)(klass))
+
+typedef struct _GstRTSPStream GstRTSPStream;
+typedef struct _GstRTSPStreamClass GstRTSPStreamClass;
+typedef struct _GstRTSPStreamPrivate GstRTSPStreamPrivate;
+
+#include "rtsp-stream-transport.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-session.h"
+
+/**
+ * GstRTSPStream:
+ *
+ * The definition of a media stream.
+ */
+struct _GstRTSPStream {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPStreamPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];};
+
+struct _GstRTSPStreamClass {
+ GObjectClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_stream_get_type (void);
+
+GstRTSPStream * gst_rtsp_stream_new (guint idx, GstElement *payloader,
+ GstPad *srcpad);
+guint gst_rtsp_stream_get_index (GstRTSPStream *stream);
+guint gst_rtsp_stream_get_pt (GstRTSPStream *stream);
+GstPad * gst_rtsp_stream_get_srcpad (GstRTSPStream *stream);
+
+void gst_rtsp_stream_set_control (GstRTSPStream *stream, const gchar *control);
+gchar * gst_rtsp_stream_get_control (GstRTSPStream *stream);
+gboolean gst_rtsp_stream_has_control (GstRTSPStream *stream, const gchar *control);
+
+void gst_rtsp_stream_set_mtu (GstRTSPStream *stream, guint mtu);
+guint gst_rtsp_stream_get_mtu (GstRTSPStream *stream);
+
+void gst_rtsp_stream_set_dscp_qos (GstRTSPStream *stream, gint dscp_qos);
+gint gst_rtsp_stream_get_dscp_qos (GstRTSPStream *stream);
+
+gboolean gst_rtsp_stream_is_transport_supported (GstRTSPStream *stream,
+ GstRTSPTransport *transport);
+
+void gst_rtsp_stream_set_profiles (GstRTSPStream *stream, GstRTSPProfile profiles);
+GstRTSPProfile gst_rtsp_stream_get_profiles (GstRTSPStream *stream);
+
+void gst_rtsp_stream_set_protocols (GstRTSPStream *stream, GstRTSPLowerTrans protocols);
+GstRTSPLowerTrans gst_rtsp_stream_get_protocols (GstRTSPStream *stream);
+
+void gst_rtsp_stream_set_address_pool (GstRTSPStream *stream, GstRTSPAddressPool *pool);
+GstRTSPAddressPool *
+ gst_rtsp_stream_get_address_pool (GstRTSPStream *stream);
+
+GstRTSPAddress * gst_rtsp_stream_reserve_address (GstRTSPStream *stream,
+ const gchar * address,
+ guint port,
+ guint n_ports,
+ guint ttl);
+
+gboolean gst_rtsp_stream_join_bin (GstRTSPStream *stream,
+ GstBin *bin, GstElement *rtpbin,
+ GstState state);
+gboolean gst_rtsp_stream_leave_bin (GstRTSPStream *stream,
+ GstBin *bin, GstElement *rtpbin);
+
+gboolean gst_rtsp_stream_set_blocked (GstRTSPStream * stream,
+ gboolean blocked);
+gboolean gst_rtsp_stream_is_blocking (GstRTSPStream * stream);
+
+void gst_rtsp_stream_get_server_port (GstRTSPStream *stream,
+ GstRTSPRange *server_port,
+ GSocketFamily family);
+GstRTSPAddress * gst_rtsp_stream_get_multicast_address (GstRTSPStream *stream,
+ GSocketFamily family);
+
+
+GObject * gst_rtsp_stream_get_rtpsession (GstRTSPStream *stream);
+
+void gst_rtsp_stream_get_ssrc (GstRTSPStream *stream,
+ guint *ssrc);
+
+gboolean gst_rtsp_stream_get_rtpinfo (GstRTSPStream *stream,
+ guint *rtptime, guint *seq,
+ guint *clock_rate,
+ GstClockTime *running_time);
+GstCaps * gst_rtsp_stream_get_caps (GstRTSPStream *stream);
+
+GstFlowReturn gst_rtsp_stream_recv_rtp (GstRTSPStream *stream,
+ GstBuffer *buffer);
+GstFlowReturn gst_rtsp_stream_recv_rtcp (GstRTSPStream *stream,
+ GstBuffer *buffer);
+
+gboolean gst_rtsp_stream_add_transport (GstRTSPStream *stream,
+ GstRTSPStreamTransport *trans);
+gboolean gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
+ GstRTSPStreamTransport *trans);
+
+GSocket * gst_rtsp_stream_get_rtp_socket (GstRTSPStream *stream,
+ GSocketFamily family);
+GSocket * gst_rtsp_stream_get_rtcp_socket (GstRTSPStream *stream,
+ GSocketFamily family);
+
+gboolean gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
+ guint ssrc, GstCaps * crypto);
+
+
+gboolean gst_rtsp_stream_query_position (GstRTSPStream * stream,
+ gint64 * position);
+gboolean gst_rtsp_stream_query_stop (GstRTSPStream * stream,
+ gint64 * stop);
+
+void gst_rtsp_stream_set_seqnum_offset (GstRTSPStream *stream, guint16 seqnum);
+guint16 gst_rtsp_stream_get_current_seqnum (GstRTSPStream *stream);
+guint64 gst_rtsp_stream_get_udp_sent_bytes (GstRTSPStream *stream);
+
+/**
+ * GstRTSPStreamTransportFilterFunc:
+ * @stream: a #GstRTSPStream object
+ * @trans: a #GstRTSPStreamTransport in @stream
+ * @user_data: user data that has been given to gst_rtsp_stream_transport_filter()
+ *
+ * This function will be called by the gst_rtsp_stream_transport_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @trans will be removed
+ * from @stream.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @trans untouched in
+ * @stream.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @trans to the result #GList of
+ * gst_rtsp_stream_transport_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPStreamTransportFilterFunc) (GstRTSPStream *stream,
+ GstRTSPStreamTransport *trans,
+ gpointer user_data);
+
+GList * gst_rtsp_stream_transport_filter (GstRTSPStream *stream,
+ GstRTSPStreamTransportFilterFunc func,
+ gpointer user_data);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_STREAM_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-thread-pool
+ * @short_description: A pool of threads
+ * @see_also: #GstRTSPMedia, #GstRTSPClient
+ *
+ * A #GstRTSPThreadPool manages reusable threads for various server tasks.
+ * Currently the defined thread types can be found in #GstRTSPThreadType.
+ *
+ * Threads of type #GST_RTSP_THREAD_TYPE_CLIENT are used to handle requests from
+ * a connected client. With gst_rtsp_thread_pool_get_max_threads() a maximum
+ * number of threads can be set after which the pool will start to reuse the
+ * same thread for multiple clients.
+ *
+ * Threads of type #GST_RTSP_THREAD_TYPE_MEDIA will be used to perform the state
+ * changes of the media pipelines and handle its bus messages.
+ *
+ * gst_rtsp_thread_pool_get_thread() can be used to create a #GstRTSPThread
+ * object of the right type. The thread object contains a mainloop and context
+ * that run in a seperate thread and can be used to attached sources to.
+ *
+ * gst_rtsp_thread_reuse() can be used to reuse a thread for multiple purposes.
+ * If all gst_rtsp_thread_reuse() calls are matched with a
+ * gst_rtsp_thread_stop() call, the mainloop will be quit and the thread will
+ * stop.
+ *
+ * To configure the threads, a subclass of this object should be made and the
+ * virtual methods should be overriden to implement the desired functionality.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include <string.h>
+
+#include "rtsp-thread-pool.h"
+
+typedef struct _GstRTSPThreadImpl
+{
+ GstRTSPThread thread;
+
+ gint reused;
+ GSource *source;
+} GstRTSPThreadImpl;
+
+GST_DEFINE_MINI_OBJECT_TYPE (GstRTSPThread, gst_rtsp_thread);
+
+static void gst_rtsp_thread_init (GstRTSPThreadImpl * impl);
+
+static void
+_gst_rtsp_thread_free (GstRTSPThreadImpl * impl)
+{
+ GST_DEBUG ("free thread %p", impl);
+
+ g_source_unref (impl->source);
+ g_main_loop_unref (impl->thread.loop);
+ g_main_context_unref (impl->thread.context);
+ g_slice_free1 (sizeof (GstRTSPThreadImpl), impl);
+}
+
+static GstRTSPThread *
+_gst_rtsp_thread_copy (GstRTSPThreadImpl * impl)
+{
+ GstRTSPThreadImpl *copy;
+
+ GST_DEBUG ("copy thread %p", impl);
+
+ copy = g_slice_new0 (GstRTSPThreadImpl);
+ gst_rtsp_thread_init (copy);
+ copy->thread.context = g_main_context_ref (impl->thread.context);
+ copy->thread.loop = g_main_loop_ref (impl->thread.loop);
+
+ return GST_RTSP_THREAD (copy);
+}
+
+static void
+gst_rtsp_thread_init (GstRTSPThreadImpl * impl)
+{
+ gst_mini_object_init (GST_MINI_OBJECT_CAST (impl), 0,
+ GST_TYPE_RTSP_THREAD,
+ (GstMiniObjectCopyFunction) _gst_rtsp_thread_copy, NULL,
+ (GstMiniObjectFreeFunction) _gst_rtsp_thread_free);
+
+ g_atomic_int_set (&impl->reused, 1);
+}
+
+/**
+ * gst_rtsp_thread_new:
+ * @type: the thread type
+ *
+ * Create a new thread object that can run a mainloop.
+ *
+ * Returns: (transfer full): a #GstRTSPThread.
+ */
+GstRTSPThread *
+gst_rtsp_thread_new (GstRTSPThreadType type)
+{
+ GstRTSPThreadImpl *impl;
+
+ impl = g_slice_new0 (GstRTSPThreadImpl);
+
+ gst_rtsp_thread_init (impl);
+ impl->thread.type = type;
+ impl->thread.context = g_main_context_new ();
+ impl->thread.loop = g_main_loop_new (impl->thread.context, TRUE);
+
+ return GST_RTSP_THREAD (impl);
+}
+
+/**
+ * gst_rtsp_thread_reuse:
+ * @thread: (transfer none): a #GstRTSPThread
+ *
+ * Reuse the mainloop of @thread
+ *
+ * Returns: %TRUE if the mainloop could be reused
+ */
+gboolean
+gst_rtsp_thread_reuse (GstRTSPThread * thread)
+{
+ GstRTSPThreadImpl *impl = (GstRTSPThreadImpl *) thread;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_THREAD (thread), FALSE);
+
+ GST_DEBUG ("reuse thread %p", thread);
+
+ res = g_atomic_int_add (&impl->reused, 1) > 0;
+ if (res)
+ gst_rtsp_thread_ref (thread);
+
+ return res;
+}
+
+static gboolean
+do_quit (GstRTSPThread * thread)
+{
+ GST_DEBUG ("stop mainloop of thread %p", thread);
+ g_main_loop_quit (thread->loop);
+ return FALSE;
+}
+
+/**
+ * gst_rtsp_thread_stop:
+ * @thread: (transfer full): a #GstRTSPThread
+ *
+ * Stop and unref @thread. When no threads are using the mainloop, the thread
+ * will be stopped and the final ref to @thread will be released.
+ */
+void
+gst_rtsp_thread_stop (GstRTSPThread * thread)
+{
+ GstRTSPThreadImpl *impl = (GstRTSPThreadImpl *) thread;
+
+ g_return_if_fail (GST_IS_RTSP_THREAD (thread));
+
+ GST_DEBUG ("stop thread %p", thread);
+
+ if (g_atomic_int_dec_and_test (&impl->reused)) {
+ GST_DEBUG ("add idle source to quit mainloop of thread %p", thread);
+ impl->source = g_idle_source_new ();
+ g_source_set_callback (impl->source, (GSourceFunc) do_quit,
+ thread, (GDestroyNotify) gst_rtsp_thread_unref);
+ g_source_attach (impl->source, thread->context);
+ } else
+ gst_rtsp_thread_unref (thread);
+}
+
+#define GST_RTSP_THREAD_POOL_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPoolPrivate))
+
+struct _GstRTSPThreadPoolPrivate
+{
+ GMutex lock;
+
+ gint max_threads;
+ /* currently used mainloops */
+ GQueue threads;
+};
+
+#define DEFAULT_MAX_THREADS 1
+
+enum
+{
+ PROP_0,
+ PROP_MAX_THREADS,
+ PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_thread_pool_debug);
+#define GST_CAT_DEFAULT rtsp_thread_pool_debug
+
+static GQuark thread_pool;
+
+static void gst_rtsp_thread_pool_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_thread_pool_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_thread_pool_finalize (GObject * obj);
+
+static gpointer do_loop (GstRTSPThread * thread);
+static GstRTSPThread *default_get_thread (GstRTSPThreadPool * pool,
+ GstRTSPThreadType type, GstRTSPContext * ctx);
+
+G_DEFINE_TYPE (GstRTSPThreadPool, gst_rtsp_thread_pool, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_thread_pool_class_init (GstRTSPThreadPoolClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ g_type_class_add_private (klass, sizeof (GstRTSPThreadPoolPrivate));
+
+ gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->get_property = gst_rtsp_thread_pool_get_property;
+ gobject_class->set_property = gst_rtsp_thread_pool_set_property;
+ gobject_class->finalize = gst_rtsp_thread_pool_finalize;
+
+ /**
+ * GstRTSPThreadPool::max-threads:
+ *
+ * The maximum amount of threads to use for client connections. A value of
+ * 0 means to use only the mainloop, -1 means an unlimited amount of
+ * threads.
+ */
+ g_object_class_install_property (gobject_class, PROP_MAX_THREADS,
+ g_param_spec_int ("max-threads", "Max Threads",
+ "The maximum amount of threads to use for client connections "
+ "(0 = only mainloop, -1 = unlimited)", -1, G_MAXINT,
+ DEFAULT_MAX_THREADS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ klass->get_thread = default_get_thread;
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_thread_pool_debug, "rtspthreadpool", 0,
+ "GstRTSPThreadPool");
+
+ thread_pool = g_quark_from_string ("gst.rtsp.thread.pool");
+}
+
+static void
+gst_rtsp_thread_pool_init (GstRTSPThreadPool * pool)
+{
+ GstRTSPThreadPoolPrivate *priv;
+
+ pool->priv = priv = GST_RTSP_THREAD_POOL_GET_PRIVATE (pool);
+
+ g_mutex_init (&priv->lock);
+ priv->max_threads = DEFAULT_MAX_THREADS;
+ g_queue_init (&priv->threads);
+}
+
+static void
+gst_rtsp_thread_pool_finalize (GObject * obj)
+{
+ GstRTSPThreadPool *pool = GST_RTSP_THREAD_POOL (obj);
+ GstRTSPThreadPoolPrivate *priv = pool->priv;
+
+ GST_INFO ("finalize pool %p", pool);
+
+ g_queue_clear (&priv->threads);
+ g_mutex_clear (&priv->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_thread_pool_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_thread_pool_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPThreadPool *pool = GST_RTSP_THREAD_POOL (object);
+
+ switch (propid) {
+ case PROP_MAX_THREADS:
+ g_value_set_int (value, gst_rtsp_thread_pool_get_max_threads (pool));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static void
+gst_rtsp_thread_pool_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPThreadPool *pool = GST_RTSP_THREAD_POOL (object);
+
+ switch (propid) {
+ case PROP_MAX_THREADS:
+ gst_rtsp_thread_pool_set_max_threads (pool, g_value_get_int (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+ }
+}
+
+static gpointer
+do_loop (GstRTSPThread * thread)
+{
+ GstRTSPThreadPoolPrivate *priv;
+ GstRTSPThreadPoolClass *klass;
+ GstRTSPThreadPool *pool;
+
+ pool = gst_mini_object_get_qdata (GST_MINI_OBJECT (thread), thread_pool);
+ priv = pool->priv;
+
+ klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+ if (klass->thread_enter)
+ klass->thread_enter (pool, thread);
+
+ GST_INFO ("enter mainloop of thread %p", thread);
+ g_main_loop_run (thread->loop);
+ GST_INFO ("exit mainloop of thread %p", thread);
+
+ if (klass->thread_leave)
+ klass->thread_leave (pool, thread);
+
+ g_mutex_lock (&priv->lock);
+ g_queue_remove (&priv->threads, thread);
+ g_mutex_unlock (&priv->lock);
+
+ gst_rtsp_thread_unref (thread);
+
+ return NULL;
+}
+
+/**
+ * gst_rtsp_thread_pool_new:
+ *
+ * Create a new #GstRTSPThreadPool instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPThreadPool
+ */
+GstRTSPThreadPool *
+gst_rtsp_thread_pool_new (void)
+{
+ GstRTSPThreadPool *result;
+
+ result = g_object_new (GST_TYPE_RTSP_THREAD_POOL, NULL);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_thread_pool_set_max_threads:
+ * @pool: a #GstRTSPThreadPool
+ * @max_threads: maximum threads
+ *
+ * Set the maximum threads used by the pool to handle client requests.
+ * A value of 0 will use the pool mainloop, a value of -1 will use an
+ * unlimited number of threads.
+ */
+void
+gst_rtsp_thread_pool_set_max_threads (GstRTSPThreadPool * pool,
+ gint max_threads)
+{
+ GstRTSPThreadPoolPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_THREAD_POOL (pool));
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->max_threads = max_threads;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_thread_pool_get_max_threads:
+ * @pool: a #GstRTSPThreadPool
+ *
+ * Get the maximum number of threads used for client connections.
+ * See gst_rtsp_thread_pool_set_max_threads().
+ *
+ * Returns: the maximum number of threads.
+ */
+gint
+gst_rtsp_thread_pool_get_max_threads (GstRTSPThreadPool * pool)
+{
+ GstRTSPThreadPoolPrivate *priv;
+ gint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_THREAD_POOL (pool), -1);
+
+ priv = pool->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->max_threads;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+static GstRTSPThread *
+make_thread (GstRTSPThreadPool * pool, GstRTSPThreadType type,
+ GstRTSPContext * ctx)
+{
+ GstRTSPThreadPoolClass *klass;
+ GstRTSPThread *thread;
+
+ klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+ thread = gst_rtsp_thread_new (type);
+ gst_mini_object_set_qdata (GST_MINI_OBJECT (thread), thread_pool,
+ g_object_ref (pool), g_object_unref);
+
+ GST_DEBUG_OBJECT (pool, "new thread %p", thread);
+
+ if (klass->configure_thread)
+ klass->configure_thread (pool, thread, ctx);
+
+ return thread;
+}
+
+static GstRTSPThread *
+default_get_thread (GstRTSPThreadPool * pool,
+ GstRTSPThreadType type, GstRTSPContext * ctx)
+{
+ GstRTSPThreadPoolPrivate *priv = pool->priv;
+ GstRTSPThreadPoolClass *klass;
+ GstRTSPThread *thread;
+ GError *error = NULL;
+
+ klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+ switch (type) {
+ case GST_RTSP_THREAD_TYPE_CLIENT:
+ if (priv->max_threads == 0) {
+ /* no threads allowed */
+ GST_DEBUG_OBJECT (pool, "no client threads allowed");
+ thread = NULL;
+ } else {
+ g_mutex_lock (&priv->lock);
+ retry:
+ if (priv->max_threads > 0 &&
+ g_queue_get_length (&priv->threads) >= priv->max_threads) {
+ /* max threads reached, recycle from queue */
+ thread = g_queue_pop_head (&priv->threads);
+ GST_DEBUG_OBJECT (pool, "recycle client thread %p", thread);
+ if (!gst_rtsp_thread_reuse (thread)) {
+ GST_DEBUG_OBJECT (pool, "thread %p stopping, retry", thread);
+ /* this can happen if we just decremented the reuse counter of the
+ * thread and signaled the mainloop that it should stop. We leave
+ * the thread out of the queue now, there is no point to add it
+ * again, it will be removed from the mainloop otherwise after it
+ * stops. */
+ goto retry;
+ }
+ } else {
+ /* make more threads */
+ GST_DEBUG_OBJECT (pool, "make new client thread");
+ thread = make_thread (pool, type, ctx);
+
+ if (!g_thread_pool_push (klass->pool, gst_rtsp_thread_ref (thread),
+ &error))
+ goto thread_error;
+ }
+ g_queue_push_tail (&priv->threads, thread);
+ g_mutex_unlock (&priv->lock);
+ }
+ break;
+ case GST_RTSP_THREAD_TYPE_MEDIA:
+ GST_DEBUG_OBJECT (pool, "make new media thread");
+ thread = make_thread (pool, type, ctx);
+
+ if (!g_thread_pool_push (klass->pool, gst_rtsp_thread_ref (thread),
+ &error))
+ goto thread_error;
+ break;
+ default:
+ thread = NULL;
+ break;
+ }
+ return thread;
+
+ /* ERRORS */
+thread_error:
+ {
+ GST_ERROR_OBJECT (pool, "failed to push thread %s", error->message);
+ gst_rtsp_thread_unref (thread);
+ /* drop also the ref dedicated for the pool */
+ gst_rtsp_thread_unref (thread);
+ g_clear_error (&error);
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_thread_pool_get_thread:
+ * @pool: a #GstRTSPThreadPool
+ * @type: the #GstRTSPThreadType
+ * @ctx: (transfer none): a #GstRTSPContext
+ *
+ * Get a new #GstRTSPThread for @type and @ctx.
+ *
+ * Returns: (transfer full): a new #GstRTSPThread, gst_rtsp_thread_stop() after usage
+ */
+GstRTSPThread *
+gst_rtsp_thread_pool_get_thread (GstRTSPThreadPool * pool,
+ GstRTSPThreadType type, GstRTSPContext * ctx)
+{
+ GstRTSPThreadPoolClass *klass;
+ GstRTSPThread *result = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_THREAD_POOL (pool), NULL);
+
+ klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+ /* We want to be thread safe as there might be 2 threads wanting to get new
+ * #GstRTSPThread at the same time
+ */
+ if (G_UNLIKELY (!g_atomic_pointer_get (&klass->pool))) {
+ GThreadPool *t_pool;
+ t_pool = g_thread_pool_new ((GFunc) do_loop, klass, -1, FALSE, NULL);
+ if (!g_atomic_pointer_compare_and_exchange (&klass->pool, NULL, t_pool))
+ g_thread_pool_free (t_pool, FALSE, TRUE);
+ }
+
+ if (klass->get_thread)
+ result = klass->get_thread (pool, type, ctx);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_thread_pool_cleanup:
+ *
+ * Wait for all tasks to be stopped and free all allocated resources. This is
+ * mainly used in test suites to ensure proper cleanup of internal data
+ * structures.
+ */
+void
+gst_rtsp_thread_pool_cleanup (void)
+{
+ GstRTSPThreadPoolClass *klass;
+
+ klass =
+ GST_RTSP_THREAD_POOL_CLASS (g_type_class_peek
+ (gst_rtsp_thread_pool_get_type ()));
+ if (klass->pool != NULL) {
+ g_thread_pool_free (klass->pool, FALSE, TRUE);
+ klass->pool = NULL;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_THREAD_POOL_H__
+#define __GST_RTSP_THREAD_POOL_H__
+
+typedef struct _GstRTSPThread GstRTSPThread;
+typedef struct _GstRTSPThreadPool GstRTSPThreadPool;
+typedef struct _GstRTSPThreadPoolClass GstRTSPThreadPoolClass;
+typedef struct _GstRTSPThreadPoolPrivate GstRTSPThreadPoolPrivate;
+
+#include "rtsp-client.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_THREAD_POOL (gst_rtsp_thread_pool_get_type ())
+#define GST_IS_RTSP_THREAD_POOL(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_THREAD_POOL))
+#define GST_IS_RTSP_THREAD_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_THREAD_POOL))
+#define GST_RTSP_THREAD_POOL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPoolClass))
+#define GST_RTSP_THREAD_POOL(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPool))
+#define GST_RTSP_THREAD_POOL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPoolClass))
+#define GST_RTSP_THREAD_POOL_CAST(obj) ((GstRTSPThreadPool*)(obj))
+#define GST_RTSP_THREAD_POOL_CLASS_CAST(klass) ((GstRTSPThreadPoolClass*)(klass))
+
+GType gst_rtsp_thread_get_type (void);
+
+#define GST_TYPE_RTSP_THREAD (gst_rtsp_thread_get_type ())
+#define GST_IS_RTSP_THREAD(obj) (GST_IS_MINI_OBJECT_TYPE (obj, GST_TYPE_RTSP_THREAD))
+#define GST_RTSP_THREAD_CAST(obj) ((GstRTSPThread*)(obj))
+#define GST_RTSP_THREAD(obj) (GST_RTSP_THREAD_CAST(obj))
+
+/**
+ * GstRTSPThreadType:
+ * @GST_RTSP_THREAD_TYPE_CLIENT: a thread to handle the client communication
+ * @GST_RTSP_THREAD_TYPE_MEDIA: a thread to handle media
+ *
+ * Different thread types
+ */
+typedef enum
+{
+ GST_RTSP_THREAD_TYPE_CLIENT,
+ GST_RTSP_THREAD_TYPE_MEDIA
+} GstRTSPThreadType;
+
+/**
+ * GstRTSPThread:
+ * @mini_object: parent #GstMiniObject
+ * @type: the thread type
+ * @context: a #GMainContext
+ * @loop: a #GMainLoop
+ *
+ * Structure holding info about a mainloop running in a thread
+ */
+struct _GstRTSPThread {
+ GstMiniObject mini_object;
+
+ GstRTSPThreadType type;
+ GMainContext *context;
+ GMainLoop *loop;
+};
+
+GstRTSPThread * gst_rtsp_thread_new (GstRTSPThreadType type);
+
+gboolean gst_rtsp_thread_reuse (GstRTSPThread * thread);
+void gst_rtsp_thread_stop (GstRTSPThread * thread);
+
+/**
+ * gst_rtsp_thread_ref:
+ * @thread: The thread to refcount
+ *
+ * Increase the refcount of this thread.
+ *
+ * Returns: (transfer full): @thread (for convenience when doing assignments)
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC GstRTSPThread * gst_rtsp_thread_ref (GstRTSPThread * thread);
+#endif
+
+static inline GstRTSPThread *
+gst_rtsp_thread_ref (GstRTSPThread * thread)
+{
+ return (GstRTSPThread *) gst_mini_object_ref (GST_MINI_OBJECT_CAST (thread));
+}
+
+/**
+ * gst_rtsp_thread_unref:
+ * @thread: (transfer full): the thread to refcount
+ *
+ * Decrease the refcount of an thread, freeing it if the refcount reaches 0.
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC void gst_rtsp_thread_unref (GstRTSPPermissions * thread);
+#endif
+
+
+static inline void
+gst_rtsp_thread_unref (GstRTSPThread * thread)
+{
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (thread));
+}
+
+/**
+ * GstRTSPThreadPool:
+ *
+ * The thread pool structure.
+ */
+struct _GstRTSPThreadPool {
+ GObject parent;
+
+ /*< private >*/
+ GstRTSPThreadPoolPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPThreadPoolClass:
+ * @pool: a #GThreadPool used internally
+ * @get_thread: this function should make or reuse an existing thread that runs
+ * a mainloop.
+ * @configure_thread: configure a thread object. this vmethod is called when
+ * a new thread has been created and should be configured.
+ * @thread_enter: called from the thread when it is entered
+ * @thread_leave: called from the thread when it is left
+ *
+ * Class for managing threads.
+ */
+struct _GstRTSPThreadPoolClass {
+ GObjectClass parent_class;
+
+ GThreadPool *pool;
+
+ GstRTSPThread * (*get_thread) (GstRTSPThreadPool *pool,
+ GstRTSPThreadType type,
+ GstRTSPContext *ctx);
+ void (*configure_thread) (GstRTSPThreadPool *pool,
+ GstRTSPThread * thread,
+ GstRTSPContext *ctx);
+
+ void (*thread_enter) (GstRTSPThreadPool *pool,
+ GstRTSPThread *thread);
+ void (*thread_leave) (GstRTSPThreadPool *pool,
+ GstRTSPThread *thread);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_thread_pool_get_type (void);
+
+GstRTSPThreadPool * gst_rtsp_thread_pool_new (void);
+
+void gst_rtsp_thread_pool_set_max_threads (GstRTSPThreadPool * pool, gint max_threads);
+gint gst_rtsp_thread_pool_get_max_threads (GstRTSPThreadPool * pool);
+
+GstRTSPThread * gst_rtsp_thread_pool_get_thread (GstRTSPThreadPool *pool,
+ GstRTSPThreadType type,
+ GstRTSPContext *ctx);
+void gst_rtsp_thread_pool_cleanup (void);
+G_END_DECLS
+
+#endif /* __GST_RTSP_THREAD_POOL_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-token
+ * @short_description: Roles and permissions for a client
+ * @see_also: #GstRTSPClient, #GstRTSPPermissions, #GstRTSPAuth
+ *
+ * A #GstRTSPToken contains the permissions and roles of the user
+ * performing the current request. A token is usually created when a user is
+ * authenticated by the #GstRTSPAuth object and is then placed as the current
+ * token for the current request.
+ *
+ * #GstRTSPAuth can use the token and its contents to check authorization for
+ * various operations by comparing the token to the #GstRTSPPermissions of the
+ * object.
+ *
+ * The accepted values of the token are entirely defined by the #GstRTSPAuth
+ * object that implements the security policy.
+ *
+ * Last reviewed on 2013-07-15 (1.0.0)
+ */
+
+#include <string.h>
+
+#include "rtsp-token.h"
+
+typedef struct _GstRTSPTokenImpl
+{
+ GstRTSPToken token;
+
+ GstStructure *structure;
+} GstRTSPTokenImpl;
+
+#define GST_RTSP_TOKEN_STRUCTURE(t) (((GstRTSPTokenImpl *)(t))->structure)
+
+//GST_DEBUG_CATEGORY_STATIC (rtsp_token_debug);
+//#define GST_CAT_DEFAULT rtsp_token_debug
+
+GST_DEFINE_MINI_OBJECT_TYPE (GstRTSPToken, gst_rtsp_token);
+
+static void gst_rtsp_token_init (GstRTSPTokenImpl * token,
+ GstStructure * structure);
+
+static void
+_gst_rtsp_token_free (GstRTSPToken * token)
+{
+ GstRTSPTokenImpl *impl = (GstRTSPTokenImpl *) token;
+
+ gst_structure_set_parent_refcount (impl->structure, NULL);
+ gst_structure_free (impl->structure);
+
+ g_slice_free1 (sizeof (GstRTSPTokenImpl), token);
+}
+
+static GstRTSPToken *
+_gst_rtsp_token_copy (GstRTSPTokenImpl * token)
+{
+ GstRTSPTokenImpl *copy;
+ GstStructure *structure;
+
+ structure = gst_structure_copy (token->structure);
+
+ copy = g_slice_new0 (GstRTSPTokenImpl);
+ gst_rtsp_token_init (copy, structure);
+
+ return (GstRTSPToken *) copy;
+}
+
+static void
+gst_rtsp_token_init (GstRTSPTokenImpl * token, GstStructure * structure)
+{
+ gst_mini_object_init (GST_MINI_OBJECT_CAST (token), 0,
+ GST_TYPE_RTSP_TOKEN,
+ (GstMiniObjectCopyFunction) _gst_rtsp_token_copy, NULL,
+ (GstMiniObjectFreeFunction) _gst_rtsp_token_free);
+
+ token->structure = structure;
+ gst_structure_set_parent_refcount (token->structure,
+ &token->token.mini_object.refcount);
+}
+
+/**
+ * gst_rtsp_token_new_empty:
+ *
+ * Create a new empty Authorization token.
+ *
+ * Returns: (transfer full): a new empty authorization token.
+ */
+GstRTSPToken *
+gst_rtsp_token_new_empty (void)
+{
+ GstRTSPTokenImpl *token;
+ GstStructure *s;
+
+ s = gst_structure_new_empty ("GstRTSPToken");
+ g_return_val_if_fail (s != NULL, NULL);
+
+ token = g_slice_new0 (GstRTSPTokenImpl);
+ gst_rtsp_token_init (token, s);
+
+ return (GstRTSPToken *) token;
+}
+
+/**
+ * gst_rtsp_token_new:
+ * @firstfield: the first fieldname
+ * @...: additional arguments
+ *
+ * Create a new Authorization token with the given fieldnames and values.
+ * Arguments are given similar to gst_structure_new().
+ *
+ * Returns: (transfer full): a new authorization token.
+ */
+GstRTSPToken *
+gst_rtsp_token_new (const gchar * firstfield, ...)
+{
+ GstRTSPToken *result;
+ va_list var_args;
+
+ va_start (var_args, firstfield);
+ result = gst_rtsp_token_new_valist (firstfield, var_args);
+ va_end (var_args);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_token_new_valist:
+ * @firstfield: the first fieldname
+ * @var_args: additional arguments
+ *
+ * Create a new Authorization token with the given fieldnames and values.
+ * Arguments are given similar to gst_structure_new_valist().
+ *
+ * Returns: (transfer full): a new authorization token.
+ */
+GstRTSPToken *
+gst_rtsp_token_new_valist (const gchar * firstfield, va_list var_args)
+{
+ GstRTSPToken *token;
+ GstStructure *s;
+
+ g_return_val_if_fail (firstfield != NULL, NULL);
+
+ token = gst_rtsp_token_new_empty ();
+ s = GST_RTSP_TOKEN_STRUCTURE (token);
+ gst_structure_set_valist (s, firstfield, var_args);
+
+ return token;
+}
+
+
+/**
+ * gst_rtsp_token_get_structure:
+ * @token: The #GstRTSPToken.
+ *
+ * Access the structure of the token.
+ *
+ * Returns: (transfer none): The structure of the token. The structure is still
+ * owned by the token, which means that you should not free it and that the
+ * pointer becomes invalid when you free the token.
+ *
+ * MT safe.
+ */
+const GstStructure *
+gst_rtsp_token_get_structure (GstRTSPToken * token)
+{
+ g_return_val_if_fail (GST_IS_RTSP_TOKEN (token), NULL);
+
+ return GST_RTSP_TOKEN_STRUCTURE (token);
+}
+
+/**
+ * gst_rtsp_token_writable_structure:
+ * @token: The #GstRTSPToken.
+ *
+ * Get a writable version of the structure.
+ *
+ * Returns: (transfer none): The structure of the token. The structure is still
+ * owned by the token, which means that you should not free it and that the
+ * pointer becomes invalid when you free the token. This function checks if
+ * @token is writable and will never return %NULL.
+ *
+ * MT safe.
+ */
+GstStructure *
+gst_rtsp_token_writable_structure (GstRTSPToken * token)
+{
+ g_return_val_if_fail (GST_IS_RTSP_TOKEN (token), NULL);
+ g_return_val_if_fail (gst_mini_object_is_writable (GST_MINI_OBJECT_CAST
+ (token)), NULL);
+
+ return GST_RTSP_TOKEN_STRUCTURE (token);
+}
+
+/**
+ * gst_rtsp_token_get_string:
+ * @token: a #GstRTSPToken
+ * @field: a field name
+ *
+ * Get the string value of @field in @token.
+ *
+ * Returns: (transfer none) (nullable): the string value of @field in
+ * @token or %NULL when @field is not defined in @token. The string
+ * becomes invalid when you free @token.
+ */
+const gchar *
+gst_rtsp_token_get_string (GstRTSPToken * token, const gchar * field)
+{
+ return gst_structure_get_string (GST_RTSP_TOKEN_STRUCTURE (token), field);
+}
+
+/**
+ * gst_rtsp_token_is_allowed:
+ * @token: a #GstRTSPToken
+ * @field: a field name
+ *
+ * Check if @token has a boolean @field and if it is set to %TRUE.
+ *
+ * Returns: %TRUE if @token has a boolean field named @field set to %TRUE.
+ */
+gboolean
+gst_rtsp_token_is_allowed (GstRTSPToken * token, const gchar * field)
+{
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_TOKEN (token), FALSE);
+ g_return_val_if_fail (field != NULL, FALSE);
+
+ if (!gst_structure_get_boolean (GST_RTSP_TOKEN_STRUCTURE (token), field,
+ &result))
+ result = FALSE;
+
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_TOKEN_H__
+#define __GST_RTSP_TOKEN_H__
+
+typedef struct _GstRTSPToken GstRTSPToken;
+
+#include "rtsp-auth.h"
+
+G_BEGIN_DECLS
+
+GType gst_rtsp_token_get_type(void);
+
+#define GST_TYPE_RTSP_TOKEN (gst_rtsp_token_get_type())
+#define GST_IS_RTSP_TOKEN(obj) (GST_IS_MINI_OBJECT_TYPE (obj, GST_TYPE_RTSP_TOKEN))
+#define GST_RTSP_TOKEN_CAST(obj) ((GstRTSPToken*)(obj))
+#define GST_RTSP_TOKEN(obj) (GST_RTSP_TOKEN_CAST(obj))
+
+/**
+ * GstRTSPToken:
+ *
+ * An opaque object used for checking authorisations.
+ * It is generated after successful authentication.
+ */
+struct _GstRTSPToken {
+ GstMiniObject mini_object;
+};
+
+/* refcounting */
+/**
+ * gst_rtsp_token_ref:
+ * @token: The token to refcount
+ *
+ * Increase the refcount of this token.
+ *
+ * Returns: (transfer full): @token (for convenience when doing assignments)
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC GstRTSPToken * gst_rtsp_token_ref (GstRTSPToken * token);
+#endif
+
+static inline GstRTSPToken *
+gst_rtsp_token_ref (GstRTSPToken * token)
+{
+ return (GstRTSPToken *) gst_mini_object_ref (GST_MINI_OBJECT_CAST (token));
+}
+
+/**
+ * gst_rtsp_token_unref:
+ * @token: (transfer full): the token to refcount
+ *
+ * Decrease the refcount of an token, freeing it if the refcount reaches 0.
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC void gst_rtsp_token_unref (GstRTSPToken * token);
+#endif
+
+static inline void
+gst_rtsp_token_unref (GstRTSPToken * token)
+{
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (token));
+}
+
+
+GstRTSPToken * gst_rtsp_token_new_empty (void);
+GstRTSPToken * gst_rtsp_token_new (const gchar * firstfield, ...);
+GstRTSPToken * gst_rtsp_token_new_valist (const gchar * firstfield, va_list var_args);
+
+const GstStructure * gst_rtsp_token_get_structure (GstRTSPToken *token);
+GstStructure * gst_rtsp_token_writable_structure (GstRTSPToken *token);
+
+const gchar * gst_rtsp_token_get_string (GstRTSPToken *token,
+ const gchar *field);
+gboolean gst_rtsp_token_is_allowed (GstRTSPToken *token,
+ const gchar *field);
+G_END_DECLS
+
+#endif /* __GST_RTSP_TOKEN_H__ */
--- /dev/null
+Name: gst-rtsp-server
+Summary: Multimedia Framework Library
+Version: 1.4.5
+Release: 3
+Group: System/Libraries
+License: LGPL-2.0+
+Source0: %{name}-%{version}.tar.gz
+Requires(post): /sbin/ldconfig
+Requires(postun): /sbin/ldconfig
+BuildRequires: pkgconfig(gstreamer-1.0)
+BuildRequires: pkgconfig(gstreamer-plugins-base-1.0)
+
+BuildRoot: %{_tmppath}/%{name}-%{version}-build
+
+%description
+
+%package devel
+Summary: Multimedia Framework RTSP server library (DEV)
+Group: Development/Libraries
+Requires: %{name} = %{version}-%{release}
+
+%description devel
+
+%package factory
+Summary: Multimedia Framework RTSP server Library (Factory)
+Group: Development/Libraries
+Requires: %{name} = %{version}-%{release}
+
+%description factory
+
+%prep
+%setup -q
+
+%build
+
+./autogen.sh
+
+CFLAGS+=" -DEXPORT_API=\"__attribute__((visibility(\\\"default\\\")))\" "; export CFLAGS
+LDFLAGS+="-Wl,--rpath=%{_prefix}/lib -Wl,--hash-style=both -Wl,--as-needed"; export LDFLAGS
+
+# always enable sdk build. This option should go away
+%configure --disable-static
+
+# Call make instruction with smp support
+make %{?jobs:-j%jobs}
+
+%install
+rm -rf %{buildroot}
+%make_install
+mkdir -p %{buildroot}/%{_datadir}/license
+cp -rf %{_builddir}/%{name}-%{version}/COPYING %{buildroot}%{_datadir}/license/%{name}
+
+%clean
+rm -rf %{buildroot}
+
+%post
+/sbin/ldconfig
+
+%postun
+/sbin/ldconfig
+
+%files
+%manifest gst-rtsp-server.manifest
+%defattr(-,root,root,-)
+%{_datadir}/license/%{name}
+%{_libdir}/*.so.*
+
+%files devel
+%defattr(-,root,root,-)
+%{_libdir}/*.so
+%{_includedir}/gstreamer-1.0/gst/rtsp-server/rtsp-*.h
+%{_includedir}/gstreamer-1.0/gst/rtsp-server/gstwfd*.h
+%{_libdir}/pkgconfig/*
--- /dev/null
+pcfiles = \
+ gstreamer-rtsp-server-@GST_API_VERSION@.pc
+
+pcfiles_uninstalled = \
+ gstreamer-rtsp-server-@GST_API_VERSION@-uninstalled.pc
+
+all-local: $(pcfiles) $(pcfiles_uninstalled)
+
+### how to generate pc files
+%-@GST_API_VERSION@.pc: %.pc
+ cp $< $@
+%-@GST_API_VERSION@-uninstalled.pc: %-uninstalled.pc
+ cp $< $@
+
+pkgconfigdir = $(libdir)/pkgconfig
+pkgconfig_DATA = $(pcfiles)
+
+EXTRA_DIST = \
+ gstreamer-rtsp-server.pc.in \
+ gstreamer-rtsp-server-uninstalled.pc.in
+CLEANFILES = $(pcfiles) $(pcfiles_uninstalled)
--- /dev/null
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+ install-dvi-am install-exec install-exec-am install-html \
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+
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+ $(REGISTRY_ENVIRONMENT) \
+ GST_PLUGIN_SYSTEM_PATH_1_0= \
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+ GST_PLUGIN_LOADING_WHITELIST="gstreamer:gst-plugins-base:gst-plugins-good:gst-plugins-bad"
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+ gst/media \
+ gst/stream \
+ gst/addresspool \
+ gst/threadpool \
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+ gst/token \
+ gst/sessionmedia \
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+ am__expect_failure=no;; \
+esac; \
+$(AM_TESTS_ENVIRONMENT) $(TESTS_ENVIRONMENT)
+# A shell command to get the names of the tests scripts with any registered
+# extension removed (i.e., equivalently, the names of the test logs, with
+# the '.log' extension removed). The result is saved in the shell variable
+# '$bases'. This honors runtime overriding of TESTS and TEST_LOGS. Sadly,
+# we cannot use something simpler, involving e.g., "$(TEST_LOGS:.log=)",
+# since that might cause problem with VPATH rewrites for suffix-less tests.
+# See also 'test-harness-vpath-rewrite.sh' and 'test-trs-basic.sh'.
+am__set_TESTS_bases = \
+ bases='$(TEST_LOGS)'; \
+ bases=`for i in $$bases; do echo $$i; done | sed 's/\.log$$//'`; \
+ bases=`echo $$bases`
+RECHECK_LOGS = $(TEST_LOGS)
+AM_RECURSIVE_TARGETS = check recheck
+TEST_SUITE_LOG = test-suite.log
+TEST_EXTENSIONS = @EXEEXT@ .test
+LOG_DRIVER = $(SHELL) $(top_srcdir)/test-driver
+LOG_COMPILE = $(LOG_COMPILER) $(AM_LOG_FLAGS) $(LOG_FLAGS)
+am__set_b = \
+ case '$@' in \
+ */*) \
+ case '$*' in \
+ */*) b='$*';; \
+ *) b=`echo '$@' | sed 's/\.log$$//'`; \
+ esac;; \
+ *) \
+ b='$*';; \
+ esac
+am__test_logs1 = $(TESTS:=.log)
+am__test_logs2 = $(am__test_logs1:@EXEEXT@.log=.log)
+TEST_LOGS = $(am__test_logs2:.test.log=.log)
+TEST_LOG_DRIVER = $(SHELL) $(top_srcdir)/test-driver
+TEST_LOG_COMPILE = $(TEST_LOG_COMPILER) $(AM_TEST_LOG_FLAGS) \
+ $(TEST_LOG_FLAGS)
+DISTFILES = $(DIST_COMMON) $(DIST_SOURCES) $(TEXINFOS) $(EXTRA_DIST)
+ACLOCAL = @ACLOCAL@
+ACLOCAL_AMFLAGS = @ACLOCAL_AMFLAGS@
+AMTAR = @AMTAR@
+AM_DEFAULT_VERBOSITY = @AM_DEFAULT_VERBOSITY@
+AR = @AR@
+AS = @AS@
+AUTOCONF = @AUTOCONF@
+AUTOHEADER = @AUTOHEADER@
+AUTOMAKE = @AUTOMAKE@
+AWK = @AWK@
+CAT_ENTRY_END = @CAT_ENTRY_END@
+CAT_ENTRY_START = @CAT_ENTRY_START@
+CC = @CC@
+CCAS = @CCAS@
+CCASDEPMODE = @CCASDEPMODE@
+CCASFLAGS = @CCASFLAGS@
+CCDEPMODE = @CCDEPMODE@
+CFLAGS = @CFLAGS@
+CPP = @CPP@
+CPPFLAGS = @CPPFLAGS@
+CXXFLAGS = @CXXFLAGS@
+CYGPATH_W = @CYGPATH_W@
+DEFAULT_AUDIOSINK = @DEFAULT_AUDIOSINK@
+DEFAULT_AUDIOSRC = @DEFAULT_AUDIOSRC@
+DEFAULT_VIDEOSINK = @DEFAULT_VIDEOSINK@
+DEFAULT_VIDEOSRC = @DEFAULT_VIDEOSRC@
+DEFAULT_VISUALIZER = @DEFAULT_VISUALIZER@
+DEFS = @DEFS@
+DEPDIR = @DEPDIR@
+DEPRECATED_CFLAGS = @DEPRECATED_CFLAGS@
+DLLTOOL = @DLLTOOL@
+DOCBOOK_ROOT = @DOCBOOK_ROOT@
+DSYMUTIL = @DSYMUTIL@
+DUMPBIN = @DUMPBIN@
+ECHO_C = @ECHO_C@
+ECHO_N = @ECHO_N@
+ECHO_T = @ECHO_T@
+EGREP = @EGREP@
+ERROR_CFLAGS = @ERROR_CFLAGS@
+EXEEXT = @EXEEXT@
+FFLAGS = @FFLAGS@
+FGREP = @FGREP@
+GCOV = @GCOV@
+GCOV_CFLAGS = @GCOV_CFLAGS@
+GCOV_LIBS = @GCOV_LIBS@
+GIO_CFLAGS = @GIO_CFLAGS@
+GIO_LDFLAGS = @GIO_LDFLAGS@
+GIO_LIBS = @GIO_LIBS@
+GLIB_CFLAGS = @GLIB_CFLAGS@
+GLIB_EXTRA_CFLAGS = @GLIB_EXTRA_CFLAGS@
+GLIB_GENMARSHAL = @GLIB_GENMARSHAL@
+GLIB_LIBS = @GLIB_LIBS@
+GLIB_MKENUMS = @GLIB_MKENUMS@
+GLIB_REQ = @GLIB_REQ@
+GREP = @GREP@
+GSTPB_PLUGINS_DIR = @GSTPB_PLUGINS_DIR@
+GSTPD_PLUGINS_DIR = @GSTPD_PLUGINS_DIR@
+GSTPG_PLUGINS_DIR = @GSTPG_PLUGINS_DIR@
+GST_AGE = @GST_AGE@
+GST_ALL_CFLAGS = @GST_ALL_CFLAGS@
+GST_ALL_LDFLAGS = @GST_ALL_LDFLAGS@
+GST_ALL_LIBS = @GST_ALL_LIBS@
+GST_API_VERSION = @GST_API_VERSION@
+GST_BASE_CFLAGS = @GST_BASE_CFLAGS@
+GST_BASE_LIBS = @GST_BASE_LIBS@
+GST_CFLAGS = @GST_CFLAGS@
+GST_CHECK_CFLAGS = @GST_CHECK_CFLAGS@
+GST_CHECK_LIBS = @GST_CHECK_LIBS@
+GST_CURRENT = @GST_CURRENT@
+GST_LEVEL_DEFAULT = @GST_LEVEL_DEFAULT@
+GST_LIBS = @GST_LIBS@
+GST_LIBVERSION = @GST_LIBVERSION@
+GST_LIB_LDFLAGS = @GST_LIB_LDFLAGS@
+GST_LICENSE = @GST_LICENSE@
+GST_LT_LDFLAGS = @GST_LT_LDFLAGS@
+GST_OBJ_CFLAGS = @GST_OBJ_CFLAGS@
+GST_OBJ_LIBS = @GST_OBJ_LIBS@
+GST_OPTION_CFLAGS = @GST_OPTION_CFLAGS@
+GST_PACKAGE_NAME = @GST_PACKAGE_NAME@
+GST_PACKAGE_ORIGIN = @GST_PACKAGE_ORIGIN@
+GST_PKG_CONFIG_PATH = @GST_PKG_CONFIG_PATH@
+GST_PLUGINS_BAD_CFLAGS = @GST_PLUGINS_BAD_CFLAGS@
+GST_PLUGINS_BAD_DIR = @GST_PLUGINS_BAD_DIR@
+GST_PLUGINS_BAD_LIBS = @GST_PLUGINS_BAD_LIBS@
+GST_PLUGINS_BASE_CFLAGS = @GST_PLUGINS_BASE_CFLAGS@
+GST_PLUGINS_BASE_DIR = @GST_PLUGINS_BASE_DIR@
+GST_PLUGINS_BASE_LIBS = @GST_PLUGINS_BASE_LIBS@
+GST_PLUGINS_DIR = @GST_PLUGINS_DIR@
+GST_PLUGINS_GOOD_CFLAGS = @GST_PLUGINS_GOOD_CFLAGS@
+GST_PLUGINS_GOOD_DIR = @GST_PLUGINS_GOOD_DIR@
+GST_PLUGINS_GOOD_LIBS = @GST_PLUGINS_GOOD_LIBS@
+GST_REVISION = @GST_REVISION@
+GST_TOOLS_DIR = @GST_TOOLS_DIR@
+GTKDOC_CHECK = @GTKDOC_CHECK@
+GTKDOC_DEPS_CFLAGS = @GTKDOC_DEPS_CFLAGS@
+GTKDOC_DEPS_LIBS = @GTKDOC_DEPS_LIBS@
+GTKDOC_MKPDF = @GTKDOC_MKPDF@
+GTKDOC_REBASE = @GTKDOC_REBASE@
+HAVE_DOCBOOK2PS = @HAVE_DOCBOOK2PS@
+HAVE_DVIPS = @HAVE_DVIPS@
+HAVE_EPSTOPDF = @HAVE_EPSTOPDF@
+HAVE_JADETEX = @HAVE_JADETEX@
+HAVE_PNGTOPNM = @HAVE_PNGTOPNM@
+HAVE_PNMTOPS = @HAVE_PNMTOPS@
+HAVE_PS2PDF = @HAVE_PS2PDF@
+HAVE_XMLLINT = @HAVE_XMLLINT@
+HAVE_XSLTPROC = @HAVE_XSLTPROC@
+HTML_DIR = @HTML_DIR@
+INSTALL = @INSTALL@
+INSTALL_DATA = @INSTALL_DATA@
+INSTALL_PROGRAM = @INSTALL_PROGRAM@
+INSTALL_SCRIPT = @INSTALL_SCRIPT@
+INSTALL_STRIP_PROGRAM = @INSTALL_STRIP_PROGRAM@
+INTROSPECTION_CFLAGS = @INTROSPECTION_CFLAGS@
+INTROSPECTION_COMPILER = @INTROSPECTION_COMPILER@
+INTROSPECTION_GENERATE = @INTROSPECTION_GENERATE@
+INTROSPECTION_GIRDIR = @INTROSPECTION_GIRDIR@
+INTROSPECTION_LIBS = @INTROSPECTION_LIBS@
+INTROSPECTION_MAKEFILE = @INTROSPECTION_MAKEFILE@
+INTROSPECTION_SCANNER = @INTROSPECTION_SCANNER@
+INTROSPECTION_TYPELIBDIR = @INTROSPECTION_TYPELIBDIR@
+LD = @LD@
+LDFLAGS = @LDFLAGS@
+LIBCGROUP_CFLAGS = @LIBCGROUP_CFLAGS@
+LIBCGROUP_LIBS = @LIBCGROUP_LIBS@
+LIBOBJS = @LIBOBJS@
+LIBS = @LIBS@
+LIBTOOL = @LIBTOOL@
+LIPO = @LIPO@
+LN_S = @LN_S@
+LTLIBOBJS = @LTLIBOBJS@
+MAINT = @MAINT@
+MAKEINFO = @MAKEINFO@
+MANIFEST_TOOL = @MANIFEST_TOOL@
+MKDIR_P = @MKDIR_P@
+NM = @NM@
+NMEDIT = @NMEDIT@
+OBJDUMP = @OBJDUMP@
+OBJEXT = @OBJEXT@
+OTOOL = @OTOOL@
+OTOOL64 = @OTOOL64@
+PACKAGE = @PACKAGE@
+PACKAGE_BUGREPORT = @PACKAGE_BUGREPORT@
+PACKAGE_NAME = @PACKAGE_NAME@
+PACKAGE_STRING = @PACKAGE_STRING@
+PACKAGE_TARNAME = @PACKAGE_TARNAME@
+PACKAGE_URL = @PACKAGE_URL@
+PACKAGE_VERSION = @PACKAGE_VERSION@
+PACKAGE_VERSION_MAJOR = @PACKAGE_VERSION_MAJOR@
+PACKAGE_VERSION_MICRO = @PACKAGE_VERSION_MICRO@
+PACKAGE_VERSION_MINOR = @PACKAGE_VERSION_MINOR@
+PACKAGE_VERSION_NANO = @PACKAGE_VERSION_NANO@
+PACKAGE_VERSION_RELEASE = @PACKAGE_VERSION_RELEASE@
+PATH_SEPARATOR = @PATH_SEPARATOR@
+PKG_CONFIG = @PKG_CONFIG@
+PLUGINDIR = @PLUGINDIR@
+PROFILE_CFLAGS = @PROFILE_CFLAGS@
+RANLIB = @RANLIB@
+SED = @SED@
+SET_MAKE = @SET_MAKE@
+SHELL = @SHELL@
+STRIP = @STRIP@
+VALGRIND_CFLAGS = @VALGRIND_CFLAGS@
+VALGRIND_LIBS = @VALGRIND_LIBS@
+VALGRIND_PATH = @VALGRIND_PATH@
+VERSION = @VERSION@
+WARNING_CFLAGS = @WARNING_CFLAGS@
+XML_CATALOG = @XML_CATALOG@
+XSLTPROC = @XSLTPROC@
+XSLTPROC_FLAGS = @XSLTPROC_FLAGS@
+abs_builddir = @abs_builddir@
+abs_srcdir = @abs_srcdir@
+abs_top_builddir = @abs_top_builddir@
+abs_top_srcdir = @abs_top_srcdir@
+ac_ct_AR = @ac_ct_AR@
+ac_ct_CC = @ac_ct_CC@
+ac_ct_DUMPBIN = @ac_ct_DUMPBIN@
+am__include = @am__include@
+am__leading_dot = @am__leading_dot@
+am__quote = @am__quote@
+am__tar = @am__tar@
+am__untar = @am__untar@
+bindir = @bindir@
+build = @build@
+build_alias = @build_alias@
+build_cpu = @build_cpu@
+build_os = @build_os@
+build_vendor = @build_vendor@
+builddir = @builddir@
+datadir = @datadir@
+datarootdir = @datarootdir@
+docdir = @docdir@
+dvidir = @dvidir@
+exec_prefix = @exec_prefix@
+host = @host@
+host_alias = @host_alias@
+host_cpu = @host_cpu@
+host_os = @host_os@
+host_vendor = @host_vendor@
+htmldir = @htmldir@
+includedir = @includedir@
+infodir = @infodir@
+install_sh = @install_sh@
+libdir = @libdir@
+libexecdir = @libexecdir@
+localedir = @localedir@
+localstatedir = @localstatedir@
+mandir = @mandir@
+mkdir_p = @mkdir_p@
+oldincludedir = @oldincludedir@
+pdfdir = @pdfdir@
+plugindir = @plugindir@
+prefix = @prefix@
+program_transform_name = @program_transform_name@
+psdir = @psdir@
+sbindir = @sbindir@
+sharedstatedir = @sharedstatedir@
+srcdir = @srcdir@
+sysconfdir = @sysconfdir@
+target = @target@
+target_alias = @target_alias@
+target_cpu = @target_cpu@
+target_os = @target_os@
+target_vendor = @target_vendor@
+top_build_prefix = @top_build_prefix@
+top_builddir = @top_builddir@
+top_srcdir = @top_srcdir@
+
+# inspect every plugin feature
+GST_INSPECT = $(GST_TOOLS_DIR)/gst-inspect-$(GST_API_VERSION)
+CHECK_REGISTRY = $(top_builddir)/tests/check/test-registry.reg
+TEST_FILES_DIRECTORY = $(top_srcdir)/tests/files
+REGISTRY_ENVIRONMENT = \
+ GST_REGISTRY_1_0=$(CHECK_REGISTRY)
+
+TESTS_ENVIRONMENT = \
+ CK_DEFAULT_TIMEOUT=120 \
+ GST_STATE_IGNORE_ELEMENTS="$(STATE_IGNORE_ELEMENTS)" \
+ $(REGISTRY_ENVIRONMENT) \
+ GST_PLUGIN_SYSTEM_PATH_1_0= \
+ GST_PLUGIN_PATH_1_0=$(GST_PLUGINS_DIR):$(GSTPB_PLUGINS_DIR):$(GSTPG_PLUGINS_DIR):$(GSTPD_PLUGINS_DIR) \
+ GST_PLUGIN_LOADING_WHITELIST="gstreamer:gst-plugins-base:gst-plugins-good:gst-plugins-bad"
+
+
+# ths core dumps of some machines have PIDs appended
+CLEANFILES = core.* test-registry.*
+TESTS = $(check_PROGRAMS)
+AM_CFLAGS = -I$(top_srcdir)/gst/rtsp-server \
+ $(GST_PLUGINS_BASE_CFLAGS) \
+ $(GST_BASE_CFLAGS) \
+ $(GIO_CFLAGS) \
+ $(GST_CFLAGS) \
+ $(GST_CHECK_CFLAGS) \
+ -DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
+ -UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
+
+AM_CXXFLAGS = $(GST_CXXFLAGS) $(GST_CHECK_CFLAGS) \
+ -DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
+ -UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
+
+LDADD = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+ $(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_API_VERSION@ \
+ -lgstrtsp-@GST_API_VERSION@ -lgstsdp-@GST_API_VERSION@ \
+ $(GST_BASE_LIBS) $(GIO_LIBS) \
+ $(GST_LIBS) $(GST_CHECK_LIBS) $(GST_RTSP_SERVER_LIBS)
+
+SUPPRESSIONS = $(top_srcdir)/common/gst.supp
+all: all-am
+
+.SUFFIXES:
+.SUFFIXES: .c .lo .log .o .obj .test .test$(EXEEXT) .trs
+$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ $(srcdir)/Makefile.am $(top_srcdir)/common/check.mak $(am__configure_deps)
+ @for dep in $?; do \
+ case '$(am__configure_deps)' in \
+ *$$dep*) \
+ ( cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh ) \
+ && { if test -f $@; then exit 0; else break; fi; }; \
+ exit 1;; \
+ esac; \
+ done; \
+ echo ' cd $(top_srcdir) && $(AUTOMAKE) --gnu tests/check/Makefile'; \
+ $(am__cd) $(top_srcdir) && \
+ $(AUTOMAKE) --gnu tests/check/Makefile
+.PRECIOUS: Makefile
+Makefile: $(srcdir)/Makefile.in $(top_builddir)/config.status
+ @case '$?' in \
+ *config.status*) \
+ cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh;; \
+ *) \
+ echo ' cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe)'; \
+ cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe);; \
+ esac;
+$(top_srcdir)/common/check.mak:
+
+$(top_builddir)/config.status: $(top_srcdir)/configure $(CONFIG_STATUS_DEPENDENCIES)
+ cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+
+$(top_srcdir)/configure: @MAINTAINER_MODE_TRUE@ $(am__configure_deps)
+ cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(ACLOCAL_M4): @MAINTAINER_MODE_TRUE@ $(am__aclocal_m4_deps)
+ cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(am__aclocal_m4_deps):
+
+clean-checkPROGRAMS:
+ @list='$(check_PROGRAMS)'; test -n "$$list" || exit 0; \
+ echo " rm -f" $$list; \
+ rm -f $$list || exit $$?; \
+ test -n "$(EXEEXT)" || exit 0; \
+ list=`for p in $$list; do echo "$$p"; done | sed 's/$(EXEEXT)$$//'`; \
+ echo " rm -f" $$list; \
+ rm -f $$list
+
+clean-noinstPROGRAMS:
+ @list='$(noinst_PROGRAMS)'; test -n "$$list" || exit 0; \
+ echo " rm -f" $$list; \
+ rm -f $$list || exit $$?; \
+ test -n "$(EXEEXT)" || exit 0; \
+ list=`for p in $$list; do echo "$$p"; done | sed 's/$(EXEEXT)$$//'`; \
+ echo " rm -f" $$list; \
+ rm -f $$list
+gst/$(am__dirstamp):
+ @$(MKDIR_P) gst
+ @: > gst/$(am__dirstamp)
+gst/$(DEPDIR)/$(am__dirstamp):
+ @$(MKDIR_P) gst/$(DEPDIR)
+ @: > gst/$(DEPDIR)/$(am__dirstamp)
+gst/addresspool.$(OBJEXT): gst/$(am__dirstamp) \
+ gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/addresspool$(EXEEXT): $(gst_addresspool_OBJECTS) $(gst_addresspool_DEPENDENCIES) $(EXTRA_gst_addresspool_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/addresspool$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_addresspool_OBJECTS) $(gst_addresspool_LDADD) $(LIBS)
+gst/client.$(OBJEXT): gst/$(am__dirstamp) \
+ gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/client$(EXEEXT): $(gst_client_OBJECTS) $(gst_client_DEPENDENCIES) $(EXTRA_gst_client_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/client$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_client_OBJECTS) $(gst_client_LDADD) $(LIBS)
+gst/media.$(OBJEXT): gst/$(am__dirstamp) gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/media$(EXEEXT): $(gst_media_OBJECTS) $(gst_media_DEPENDENCIES) $(EXTRA_gst_media_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/media$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_media_OBJECTS) $(gst_media_LDADD) $(LIBS)
+gst/mediafactory.$(OBJEXT): gst/$(am__dirstamp) \
+ gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/mediafactory$(EXEEXT): $(gst_mediafactory_OBJECTS) $(gst_mediafactory_DEPENDENCIES) $(EXTRA_gst_mediafactory_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/mediafactory$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_mediafactory_OBJECTS) $(gst_mediafactory_LDADD) $(LIBS)
+gst/mountpoints.$(OBJEXT): gst/$(am__dirstamp) \
+ gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/mountpoints$(EXEEXT): $(gst_mountpoints_OBJECTS) $(gst_mountpoints_DEPENDENCIES) $(EXTRA_gst_mountpoints_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/mountpoints$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_mountpoints_OBJECTS) $(gst_mountpoints_LDADD) $(LIBS)
+gst/permissions.$(OBJEXT): gst/$(am__dirstamp) \
+ gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/permissions$(EXEEXT): $(gst_permissions_OBJECTS) $(gst_permissions_DEPENDENCIES) $(EXTRA_gst_permissions_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/permissions$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_permissions_OBJECTS) $(gst_permissions_LDADD) $(LIBS)
+gst/rtspserver.$(OBJEXT): gst/$(am__dirstamp) \
+ gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/rtspserver$(EXEEXT): $(gst_rtspserver_OBJECTS) $(gst_rtspserver_DEPENDENCIES) $(EXTRA_gst_rtspserver_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/rtspserver$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_rtspserver_OBJECTS) $(gst_rtspserver_LDADD) $(LIBS)
+gst/sessionmedia.$(OBJEXT): gst/$(am__dirstamp) \
+ gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/sessionmedia$(EXEEXT): $(gst_sessionmedia_OBJECTS) $(gst_sessionmedia_DEPENDENCIES) $(EXTRA_gst_sessionmedia_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/sessionmedia$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_sessionmedia_OBJECTS) $(gst_sessionmedia_LDADD) $(LIBS)
+gst/sessionpool.$(OBJEXT): gst/$(am__dirstamp) \
+ gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/sessionpool$(EXEEXT): $(gst_sessionpool_OBJECTS) $(gst_sessionpool_DEPENDENCIES) $(EXTRA_gst_sessionpool_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/sessionpool$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_sessionpool_OBJECTS) $(gst_sessionpool_LDADD) $(LIBS)
+gst/stream.$(OBJEXT): gst/$(am__dirstamp) \
+ gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/stream$(EXEEXT): $(gst_stream_OBJECTS) $(gst_stream_DEPENDENCIES) $(EXTRA_gst_stream_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/stream$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_stream_OBJECTS) $(gst_stream_LDADD) $(LIBS)
+gst/threadpool.$(OBJEXT): gst/$(am__dirstamp) \
+ gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/threadpool$(EXEEXT): $(gst_threadpool_OBJECTS) $(gst_threadpool_DEPENDENCIES) $(EXTRA_gst_threadpool_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/threadpool$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_threadpool_OBJECTS) $(gst_threadpool_LDADD) $(LIBS)
+gst/token.$(OBJEXT): gst/$(am__dirstamp) gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/token$(EXEEXT): $(gst_token_OBJECTS) $(gst_token_DEPENDENCIES) $(EXTRA_gst_token_DEPENDENCIES) gst/$(am__dirstamp)
+ @rm -f gst/token$(EXEEXT)
+ $(AM_V_CCLD)$(LINK) $(gst_token_OBJECTS) $(gst_token_LDADD) $(LIBS)
+
+mostlyclean-compile:
+ -rm -f *.$(OBJEXT)
+ -rm -f gst/*.$(OBJEXT)
+
+distclean-compile:
+ -rm -f *.tab.c
+
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/addresspool.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/client.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/media.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/mediafactory.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/mountpoints.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/permissions.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/rtspserver.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/sessionmedia.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/sessionpool.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/stream.Po@am__quote@
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+
+# keep target around, since it's referenced in the modules' Makefiles
+clean-local-check:
+ @echo
+
+# hangs spectacularly on some machines, so let's not do this by default yet
+@HAVE_VALGRIND_TRUE@check-valgrind:
+@HAVE_VALGRIND_TRUE@ $(MAKE) valgrind
+@HAVE_VALGRIND_FALSE@check-valgrind:
+@HAVE_VALGRIND_FALSE@ @true
+
+LOOPS ?= 10
+
+# run any given test by running make test.check
+# if the test fails, run it again at at least debug level 2
+%.check: %
+ @$(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=20 \
+ $* || \
+ $(TESTS_ENVIRONMENT) \
+ GST_DEBUG=$$GST_DEBUG,*:2 \
+ CK_DEFAULT_TIMEOUT=20 \
+ $*
+
+# just like 'check', but don't run it again if it fails (useful for debugging)
+%.check-norepeat: %
+ @$(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=20 \
+ $*
+
+# run any given test in a loop
+%.torture: %
+ @for i in `seq 1 $(LOOPS)`; do \
+ $(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=20 \
+ $*; done
+
+# run any given test in an infinite loop
+%.forever: %
+ @while true; do \
+ $(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=20 \
+ $* || break; done
+
+# valgrind any given test by running make test.valgrind
+%.valgrind: %
+ @$(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=360 \
+ G_SLICE=always-malloc \
+ $(LIBTOOL) --mode=execute \
+ $(VALGRIND_PATH) -q \
+ $(foreach s,$(SUPPRESSIONS),--suppressions=$(s)) \
+ --tool=memcheck --leak-check=full --trace-children=yes \
+ --show-possibly-lost=no \
+ --leak-resolution=high --num-callers=20 \
+ ./$* 2>&1 | tee valgrind.log
+ @if grep "==" valgrind.log > /dev/null 2>&1; then \
+ rm valgrind.log; \
+ exit 1; \
+ fi
+ @rm valgrind.log
+
+# valgrind any given test and generate suppressions for it
+%.valgrind.gen-suppressions: %
+ @$(TESTS_ENVIRONMENT) \
+ CK_DEFAULT_TIMEOUT=360 \
+ G_SLICE=always-malloc \
+ $(LIBTOOL) --mode=execute \
+ $(VALGRIND_PATH) -q \
+ $(foreach s,$(SUPPRESSIONS),--suppressions=$(s)) \
+ --tool=memcheck --leak-check=full --trace-children=yes \
+ --show-possibly-lost=no \
+ --leak-resolution=high --num-callers=20 \
+ --gen-suppressions=all \
+ ./$* 2>&1 | tee suppressions.log
+
+# valgrind torture any given test
+%.valgrind-torture: %
+ @for i in `seq 1 $(LOOPS)`; do \
+ $(MAKE) $*.valgrind || \
+ (echo "Failure after $$i runs"; exit 1) || \
+ exit 1; \
+ done
+ @banner="All $(LOOPS) loops passed"; \
+ dashes=`echo "$$banner" | sed s/./=/g`; \
+ echo $$dashes; echo $$banner; echo $$dashes
+
+# valgrind any given test until failure by running make test.valgrind-forever
+%.valgrind-forever: %
+ @while $(MAKE) $*.valgrind; do \
+ true; done
+
+# gdb any given test by running make test.gdb
+%.gdb: %
+ @$(TESTS_ENVIRONMENT) \
+ CK_FORK=no \
+ $(LIBTOOL) --mode=execute \
+ gdb $*
+
+%.lcov-reset:
+ $(MAKE) $*.lcov-run
+ $(MAKE) $*.lcov-report
+
+%.lcov: %
+ $(MAKE) $*.lcov-reset
+
+@GST_GCOV_ENABLED_TRUE@%.lcov-clean:
+@GST_GCOV_ENABLED_TRUE@ $(MAKE) -C $(top_builddir) lcov-clean
+
+@GST_GCOV_ENABLED_TRUE@%.lcov-run:
+@GST_GCOV_ENABLED_TRUE@ $(MAKE) $*.lcov-clean
+@GST_GCOV_ENABLED_TRUE@ $(MAKE) $*.check
+
+@GST_GCOV_ENABLED_TRUE@%.lcov-report:
+@GST_GCOV_ENABLED_TRUE@ $(MAKE) -C $(top_builddir) lcov-report
+@GST_GCOV_ENABLED_FALSE@%.lcov-run:
+@GST_GCOV_ENABLED_FALSE@ echo "Need to reconfigure with --enable-gcov"
+
+@GST_GCOV_ENABLED_FALSE@%.lcov-report:
+@GST_GCOV_ENABLED_FALSE@ echo "Need to reconfigure with --enable-gcov"
+
+# torture tests
+torture: $(TESTS)
+ -rm test-registry.*
+ @echo "Torturing tests ..."
+ @for i in `seq 1 $(LOOPS)`; do \
+ $(MAKE) check || \
+ (echo "Failure after $$i runs"; exit 1) || \
+ exit 1; \
+ done
+ @banner="All $(LOOPS) loops passed"; \
+ dashes=`echo "$$banner" | sed s/./=/g`; \
+ echo $$dashes; echo $$banner; echo $$dashes
+
+# forever tests
+forever: $(TESTS)
+ -rm test-registry.*
+ @echo "Forever tests ..."
+ @while true; do \
+ $(MAKE) check || \
+ (echo "Failure"; exit 1) || \
+ exit 1; \
+ done
+
+# valgrind all tests
+valgrind: $(TESTS)
+ @echo "Valgrinding tests ..."
+ @failed=0; \
+ for t in $(filter-out $(VALGRIND_TESTS_DISABLE),$(TESTS)); do \
+ $(MAKE) $$t.valgrind; \
+ if test "$$?" -ne 0; then \
+ echo "Valgrind error for test $$t"; \
+ failed=`expr $$failed + 1`; \
+ whicht="$$whicht $$t"; \
+ fi; \
+ done; \
+ if test "$$failed" -ne 0; then \
+ echo "$$failed tests had leaks or errors under valgrind:"; \
+ echo "$$whicht"; \
+ false; \
+ fi
+
+# valgrind all tests until failure
+valgrind-forever: $(TESTS)
+ -rm test-registry.*
+ @echo "Forever valgrinding tests ..."
+ @while true; do \
+ $(MAKE) valgrind || \
+ (echo "Failure"; exit 1) || \
+ exit 1; \
+ done
+
+# valgrind torture all tests
+valgrind-torture: $(TESTS)
+ -rm test-registry.*
+ @echo "Torturing and valgrinding tests ..."
+ @for i in `seq 1 $(LOOPS)`; do \
+ $(MAKE) valgrind || \
+ (echo "Failure after $$i runs"; exit 1) || \
+ exit 1; \
+ done
+ @banner="All $(LOOPS) loops passed"; \
+ dashes=`echo "$$banner" | sed s/./=/g`; \
+ echo $$dashes; echo $$banner; echo $$dashes
+
+# valgrind all tests and generate suppressions
+valgrind.gen-suppressions: $(TESTS)
+ @echo "Valgrinding tests ..."
+ @failed=0; \
+ for t in $(filter-out $(VALGRIND_TESTS_DISABLE),$(TESTS)); do \
+ $(MAKE) $$t.valgrind.gen-suppressions; \
+ if test "$$?" -ne 0; then \
+ echo "Valgrind error for test $$t"; \
+ failed=`expr $$failed + 1`; \
+ whicht="$$whicht $$t"; \
+ fi; \
+ done; \
+ if test "$$failed" -ne 0; then \
+ echo "$$failed tests had leaks or errors under valgrind:"; \
+ echo "$$whicht"; \
+ false; \
+ fi
+inspect:
+ @echo "Inspecting features ..."
+ @for e in `$(TESTS_ENVIRONMENT) $(GST_INSPECT) | head -n -2 \
+ | cut -d: -f2`; \
+ do echo Inspecting $$e; \
+ $(GST_INSPECT) $$e > /dev/null 2>&1; done
+
+help:
+ @echo
+ @echo "make check -- run all checks"
+ @echo "make torture -- run all checks $(LOOPS) times"
+ @echo "make (dir)/(test).check -- run the given check once, repeat with GST_DEBUG=*:2 if it fails"
+ @echo "make (dir)/(test).check-norepeat -- run the given check once, but don't run it again if it fails"
+ @echo "make (dir)/(test).forever -- run the given check forever"
+ @echo "make (dir)/(test).torture -- run the given check $(LOOPS) times"
+ @echo
+ @echo "make (dir)/(test).gdb -- start up gdb for the given test"
+ @echo
+ @echo "make valgrind -- valgrind all tests"
+ @echo "make valgrind-forever -- valgrind all tests forever"
+ @echo "make valgrind-torture -- valgrind all tests $(LOOPS) times"
+ @echo "make valgrind.gen-suppressions -- generate suppressions for all tests"
+ @echo " and save to suppressions.log"
+ @echo "make (dir)/(test).valgrind -- valgrind the given test"
+ @echo "make (dir)/(test).valgrind-forever -- valgrind the given test forever"
+ @echo "make (dir)/(test).valgrind-torture -- valgrind the given test $(LOOPS) times"
+ @echo "make (dir)/(test).valgrind.gen-suppressions -- generate suppressions"
+ @echo " and save to suppressions.log"
+ @echo "make inspect -- inspect all plugin features"
+ @echo
+ @echo
+ @echo "Additionally, you can use the GST_CHECKS environment variable to"
+ @echo "specify which test(s) should be run. This is useful if you are"
+ @echo "debugging a failure in one particular test, or want to reproduce"
+ @echo "a race condition in a single test."
+ @echo
+ @echo "Examples:"
+ @echo
+ @echo " GST_CHECKS=test_this,test_that make element/foobar.check"
+ @echo " GST_CHECKS=test_many_threads make element/foobar.forever"
+ @echo
+
+clean-local: clean-local-check
+
+$(CHECK_REGISTRY):
+ $(TESTS_ENVIRONMENT)
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-address-pool.h>
+
+GST_START_TEST (test_pool)
+{
+ GstRTSPAddressPool *pool;
+ GstRTSPAddress *addr, *addr2, *addr3;
+ GstRTSPAddressPoolResult res;
+
+ pool = gst_rtsp_address_pool_new ();
+
+ fail_if (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "233.252.0.0", 5000, 5010, 1));
+ fail_if (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "::1", 5000, 5010, 1));
+ fail_if (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "ff02::1", 5000, 5010, 1));
+ fail_if (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1.1", "233.252.0.1", 5000, 5010, 1));
+ fail_if (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "233.252.0.1.1", 5000, 5010, 1));
+ ASSERT_CRITICAL (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.0", "233.252.0.1", 5010, 5000, 1));
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.0", "233.252.0.255", 5000, 5010, 1));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.255.0.0", "233.255.0.0", 5000, 5010, 1));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.255.0.0", "233.255.0.0", 5020, 5020, 1));
+
+ /* should fail, we can't allocate a block of 256 ports */
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_MULTICAST, 256);
+ fail_unless (addr == NULL);
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr != NULL);
+
+ addr2 = gst_rtsp_address_copy (addr);
+
+ gst_rtsp_address_free (addr2);
+ gst_rtsp_address_free (addr);
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_MULTICAST, 4);
+ fail_unless (addr != NULL);
+
+ /* Will fail because pool is NULL */
+ ASSERT_CRITICAL (gst_rtsp_address_pool_clear (NULL));
+
+ /* will fail because an address is allocated */
+ ASSERT_CRITICAL (gst_rtsp_address_pool_clear (pool));
+
+ gst_rtsp_address_free (addr);
+
+ gst_rtsp_address_pool_clear (pool);
+
+ /* start with odd port to make sure we are allocated address
+ * starting with even port
+ */
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "FF11:DB8::1", "FF11:DB8::1", 5001, 5003, 1));
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_EVEN_PORT |
+ GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5002);
+ fail_unless (!g_ascii_strcasecmp (addr->address, "FF11:DB8::1"));
+
+ /* Will fail becuse there is only one IPv6 address left */
+ addr2 = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr2 == NULL);
+
+ /* Will fail because the only IPv6 address left has an odd port */
+ addr2 = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_EVEN_PORT |
+ GST_RTSP_ADDRESS_FLAG_MULTICAST, 1);
+ fail_unless (addr2 == NULL);
+
+ addr2 = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_IPV4 | GST_RTSP_ADDRESS_FLAG_MULTICAST, 1);
+ fail_unless (addr2 == NULL);
+
+ gst_rtsp_address_free (addr);
+
+ gst_rtsp_address_pool_clear (pool);
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.0", "233.252.0.255", 5000, 5002, 1));
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5000);
+ fail_unless (!strcmp (addr->address, "233.252.0.0"));
+
+ addr2 = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr2 != NULL);
+ fail_unless (addr2->port == 5000);
+ fail_unless (!strcmp (addr2->address, "233.252.0.1"));
+
+ gst_rtsp_address_free (addr);
+ gst_rtsp_address_free (addr2);
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_MULTICAST, 1);
+ fail_unless (addr == NULL);
+
+ gst_rtsp_address_pool_clear (pool);
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.1.1", "233.252.1.1", 5000, 5001, 1));
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 3,
+ 1, &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.2", 5000, 2,
+ 1, &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 500, 2, 1,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+ 2, &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "2000::1", 5000, 2, 2,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_EINVAL);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "ff02::1", 5000, 2, 2,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "1.1", 5000, 2, 2, &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_EINVAL);
+ fail_unless (addr == NULL);
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+ 1, &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_OK);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5000);
+ fail_unless (!strcmp (addr->address, "233.252.1.1"));
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+ 1, &addr2);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERESERVED);
+ fail_unless (addr2 == NULL);
+
+ gst_rtsp_address_free (addr);
+ gst_rtsp_address_pool_clear (pool);
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.1.1", "233.252.1.3", 5000, 5001, 1));
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+ 1, &addr);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5000);
+ fail_unless (!strcmp (addr->address, "233.252.1.1"));
+
+ res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.3", 5000, 2,
+ 1, &addr2);
+ fail_unless (addr2 != NULL);
+ fail_unless (addr2->port == 5000);
+ fail_unless (!strcmp (addr2->address, "233.252.1.3"));
+
+ addr3 = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr3 != NULL);
+ fail_unless (addr3->port == 5000);
+ fail_unless (!strcmp (addr3->address, "233.252.1.2"));
+
+ fail_unless (gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2)
+ == NULL);
+
+ gst_rtsp_address_free (addr);
+ gst_rtsp_address_free (addr2);
+ gst_rtsp_address_free (addr3);
+ gst_rtsp_address_pool_clear (pool);
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.1.1", "233.252.1.1", 5000, 5001, 1));
+ fail_if (gst_rtsp_address_pool_has_unicast_addresses (pool));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "192.168.1.1", "192.168.1.1", 6000, 6001, 0));
+ fail_unless (gst_rtsp_address_pool_has_unicast_addresses (pool));
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5000);
+ fail_unless (!strcmp (addr->address, "233.252.1.1"));
+ gst_rtsp_address_free (addr);
+
+ addr = gst_rtsp_address_pool_acquire_address (pool,
+ GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST, 2);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 6000);
+ fail_unless (!strcmp (addr->address, "192.168.1.1"));
+ gst_rtsp_address_free (addr);
+
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ GST_RTSP_ADDRESS_POOL_ANY_IPV4, GST_RTSP_ADDRESS_POOL_ANY_IPV4, 5000,
+ 5001, 0));
+ res =
+ gst_rtsp_address_pool_reserve_address (pool, "192.168.0.1", 5000, 1, 0,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+ res =
+ gst_rtsp_address_pool_reserve_address (pool, "0.0.0.0", 5000, 1, 0,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_OK);
+ gst_rtsp_address_free (addr);
+ gst_rtsp_address_pool_clear (pool);
+
+ /* Error case 2. Using ANY as min address makes it possible to allocate the
+ * same address twice */
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ GST_RTSP_ADDRESS_POOL_ANY_IPV4, "255.255.255.255", 5000, 5001, 0));
+ res =
+ gst_rtsp_address_pool_reserve_address (pool, "192.168.0.1", 5000, 1, 0,
+ &addr);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_OK);
+ res =
+ gst_rtsp_address_pool_reserve_address (pool, "192.168.0.1", 5000, 1, 0,
+ &addr2);
+ fail_unless (res == GST_RTSP_ADDRESS_POOL_ERESERVED);
+ gst_rtsp_address_free (addr);
+ gst_rtsp_address_pool_clear (pool);
+
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspaddresspool_suite (void)
+{
+ Suite *s = suite_create ("rtspaddresspool");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_pool);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspaddresspool);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-client.h>
+
+static gchar * session_id;
+static gint cseq;
+static guint expected_session_timeout = 60;
+
+static gboolean
+test_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OK);
+ fail_unless (g_str_equal (reason, "OK"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static gboolean
+test_response_400 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_BAD_REQUEST);
+ fail_unless (g_str_equal (reason, "Bad Request"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static gboolean
+test_response_404 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_NOT_FOUND);
+ fail_unless (g_str_equal (reason, "Not Found"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static gboolean
+test_response_454 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_SESSION_NOT_FOUND);
+ fail_unless (g_str_equal (reason, "Session Not Found"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static GstRTSPClient *
+setup_client (const gchar * launch_line)
+{
+ GstRTSPClient *client;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPMountPoints *mount_points;
+ GstRTSPMediaFactory *factory;
+ GstRTSPThreadPool *thread_pool;
+
+ client = gst_rtsp_client_new ();
+
+ session_pool = gst_rtsp_session_pool_new ();
+ gst_rtsp_client_set_session_pool (client, session_pool);
+
+ mount_points = gst_rtsp_mount_points_new ();
+ factory = gst_rtsp_media_factory_new ();
+ if (launch_line == NULL)
+ gst_rtsp_media_factory_set_launch (factory,
+ "videotestsrc ! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96");
+ else
+ gst_rtsp_media_factory_set_launch (factory, launch_line);
+
+ gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
+ gst_rtsp_client_set_mount_points (client, mount_points);
+
+ thread_pool = gst_rtsp_thread_pool_new ();
+ gst_rtsp_client_set_thread_pool (client, thread_pool);
+
+ g_object_unref (mount_points);
+ g_object_unref (session_pool);
+ g_object_unref (thread_pool);
+
+ return client;
+}
+
+static void
+teardown_client (GstRTSPClient * client)
+{
+ gst_rtsp_client_set_thread_pool (client, NULL);
+ g_object_unref (client);
+}
+
+GST_START_TEST (test_request)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPConnection *conn;
+ GSocket *sock;
+ GError *error = NULL;
+
+ client = gst_rtsp_client_new ();
+
+ /* OPTIONS with invalid url */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "foopy://padoop/") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+
+ gst_rtsp_message_unset (&request);
+
+ /* OPTIONS with unknown session id */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, "foobar");
+
+ gst_rtsp_client_set_send_func (client, test_response_454, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+
+ gst_rtsp_message_unset (&request);
+
+ /* OPTIONS with an absolute path instead of an absolute url */
+ /* set host information */
+ sock = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_STREAM,
+ G_SOCKET_PROTOCOL_TCP, &error);
+ g_assert_no_error (error);
+ gst_rtsp_connection_create_from_socket (sock, "localhost", 444, NULL, &conn);
+ fail_unless (gst_rtsp_client_set_connection (client, conn));
+ g_object_unref (sock);
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* OPTIONS with an absolute path instead of an absolute url with invalid
+ * host information */
+ g_object_unref (client);
+ client = gst_rtsp_client_new ();
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ g_object_unref (client);
+}
+
+GST_END_TEST;
+
+static gboolean
+test_option_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *str;
+ GstRTSPMethod methods;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OK);
+ fail_unless (g_str_equal (reason, "OK"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+ 0) == GST_RTSP_OK);
+ fail_unless (atoi (str) == cseq++);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_PUBLIC, &str,
+ 0) == GST_RTSP_OK);
+
+ methods = gst_rtsp_options_from_text (str);
+ fail_if (methods == 0);
+ fail_unless (methods == (GST_RTSP_DESCRIBE |
+ GST_RTSP_OPTIONS |
+ GST_RTSP_PAUSE |
+ GST_RTSP_PLAY |
+ GST_RTSP_SETUP |
+ GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN));
+
+ return TRUE;
+}
+
+GST_START_TEST (test_options)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = gst_rtsp_client_new ();
+
+ /* simple OPTIONS */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_option_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ g_object_unref (client);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_describe)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = gst_rtsp_client_new ();
+
+ /* simple DESCRIBE for non-existing url */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_404, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ g_object_unref (client);
+
+ /* simple DESCRIBE for an existing url */
+ client = setup_client (NULL);
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+static const gchar *expected_transport = NULL;;
+
+static gboolean
+test_setup_response_200_multicast (GstRTSPClient * client,
+ GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *str;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPSession *session;
+ gchar **session_hdr_params;
+
+ fail_unless (expected_transport != NULL);
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OK);
+ fail_unless (g_str_equal (reason, "OK"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+ 0) == GST_RTSP_OK);
+ fail_unless (atoi (str) == cseq++);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT,
+ &str, 0) == GST_RTSP_OK);
+
+ fail_unless (!strcmp (str, expected_transport));
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION,
+ &str, 0) == GST_RTSP_OK);
+ session_hdr_params = g_strsplit (str, ";", -1);
+
+ /* session-id value */
+ fail_unless (session_hdr_params[0] != NULL);
+
+ if (expected_session_timeout != 60) {
+ /* session timeout param */
+ gchar *timeout_str = g_strdup_printf ("timeout=%u",
+ expected_session_timeout);
+
+ fail_unless (session_hdr_params[1] != NULL);
+ g_strstrip (session_hdr_params[1]);
+ fail_unless (g_strcmp0 (session_hdr_params[1], timeout_str) == 0);
+ g_free (timeout_str);
+ }
+
+ session_pool = gst_rtsp_client_get_session_pool (client);
+ fail_unless (session_pool != NULL);
+
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1);
+ session = gst_rtsp_session_pool_find (session_pool, session_hdr_params[0]);
+ g_strfreev (session_hdr_params);
+
+ /* remember session id to be able to send teardown */
+ session_id = g_strdup (gst_rtsp_session_get_sessionid (session));
+ fail_unless (session_id != NULL);
+
+ fail_unless (session != NULL);
+ g_object_unref (session);
+
+ g_object_unref (session_pool);
+
+
+ return TRUE;
+}
+
+static gboolean
+test_teardown_response_200 (GstRTSPClient * client,
+ GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_OK);
+ fail_unless (g_str_equal (reason, "OK"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ return TRUE;
+}
+
+static void
+send_teardown (GstRTSPClient * client)
+{
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ fail_unless (session_id != NULL);
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION,
+ session_id);
+ gst_rtsp_client_set_send_func (client, test_teardown_response_200,
+ NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ g_free (session_id);
+ session_id = NULL;
+}
+
+static GstRTSPClient *
+setup_multicast_client (void)
+{
+ GstRTSPClient *client;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPMountPoints *mount_points;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAddressPool *address_pool;
+ GstRTSPThreadPool *thread_pool;
+
+ client = gst_rtsp_client_new ();
+
+ session_pool = gst_rtsp_session_pool_new ();
+ gst_rtsp_client_set_session_pool (client, session_pool);
+
+ mount_points = gst_rtsp_mount_points_new ();
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
+ address_pool = gst_rtsp_address_pool_new ();
+ fail_unless (gst_rtsp_address_pool_add_range (address_pool,
+ "233.252.0.1", "233.252.0.1", 5000, 5010, 1));
+ gst_rtsp_media_factory_set_address_pool (factory, address_pool);
+ gst_rtsp_media_factory_add_role (factory, "user",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
+ gst_rtsp_client_set_mount_points (client, mount_points);
+
+ thread_pool = gst_rtsp_thread_pool_new ();
+ gst_rtsp_client_set_thread_pool (client, thread_pool);
+
+ g_object_unref (mount_points);
+ g_object_unref (session_pool);
+ g_object_unref (address_pool);
+ g_object_unref (thread_pool);
+
+ return client;
+}
+
+GST_START_TEST (test_client_multicast_transport_404)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_multicast_client ();
+
+ /* simple SETUP for non-existing url */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test2/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast");
+
+ gst_rtsp_client_set_send_func (client, test_response_404, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+static void
+new_session_cb (GObject * client, GstRTSPSession * session, gpointer user_data)
+{
+ GST_DEBUG ("%p: new session %p", client, session);
+ gst_rtsp_session_set_timeout (session, expected_session_timeout);
+}
+
+GST_START_TEST (test_client_multicast_transport)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_multicast_client ();
+
+ expected_session_timeout = 20;
+ g_signal_connect (G_OBJECT (client), "new-session",
+ G_CALLBACK (new_session_cb), NULL);
+
+ /* simple SETUP with a valid URI and multicast */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast");
+
+ expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
+ NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+ expected_session_timeout = 60;
+
+ send_teardown (client);
+
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_multicast_ignore_transport_specific)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_multicast_client ();
+
+ /* simple SETUP with a valid URI and multicast and a specific dest,
+ * but ignore it */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast;destination=233.252.0.2;ttl=2;port=5001-5006;");
+
+ expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
+ NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+
+ send_teardown (client);
+
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+static gboolean
+test_setup_response_461 (GstRTSPClient * client,
+ GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *str;
+
+ fail_unless (expected_transport == NULL);
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_UNSUPPORTED_TRANSPORT);
+ fail_unless (g_str_equal (reason, "Unsupported transport"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+ 0) == GST_RTSP_OK);
+ fail_unless (atoi (str) == cseq++);
+
+
+ return TRUE;
+}
+
+GST_START_TEST (test_client_multicast_invalid_transport_specific)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPContext ctx = { NULL };
+
+ client = setup_multicast_client ();
+
+ ctx.client = client;
+ ctx.auth = gst_rtsp_auth_new ();
+ ctx.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+ G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx);
+
+ /* simple SETUP with a valid URI and multicast, but an invalid ip */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast;destination=233.252.0.2;ttl=1;port=5000-5001;");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ session_pool = gst_rtsp_client_get_session_pool (client);
+ fail_unless (session_pool != NULL);
+ /* FIXME: There seems to be a leak of a session here ! */
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
+ g_object_unref (session_pool);
+
+
+
+ /* simple SETUP with a valid URI and multicast, but an invalid prt */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast;destination=233.252.0.1;ttl=1;port=6000-6001;");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ session_pool = gst_rtsp_client_get_session_pool (client);
+ fail_unless (session_pool != NULL);
+ /* FIXME: There seems to be a leak of a session here ! */
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
+ g_object_unref (session_pool);
+
+
+
+ /* simple SETUP with a valid URI and multicast, but an invalid ttl */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast;destination=233.252.0.1;ttl=2;port=5000-5001;");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ session_pool = gst_rtsp_client_get_session_pool (client);
+ fail_unless (session_pool != NULL);
+ /* FIXME: There seems to be a leak of a session here ! */
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
+ g_object_unref (session_pool);
+
+
+ teardown_client (client);
+ g_object_unref (ctx.auth);
+ gst_rtsp_token_unref (ctx.token);
+ gst_rtsp_context_pop_current (&ctx);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_multicast_transport_specific)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPContext ctx = { NULL };
+
+ client = setup_multicast_client ();
+
+ ctx.client = client;
+ ctx.auth = gst_rtsp_auth_new ();
+ ctx.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+ G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx);
+
+ expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+
+ /* simple SETUP with a valid URI and multicast, but an invalid ip */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ expected_transport);
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
+ NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ expected_transport = NULL;
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
+ NULL, NULL);
+ session_pool = gst_rtsp_client_get_session_pool (client);
+ fail_unless (session_pool != NULL);
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1);
+ g_object_unref (session_pool);
+
+ send_teardown (client);
+
+ teardown_client (client);
+ g_object_unref (ctx.auth);
+ gst_rtsp_token_unref (ctx.token);
+ gst_rtsp_context_pop_current (&ctx);
+}
+
+GST_END_TEST;
+
+static gboolean
+test_response_sdp (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
+{
+ guint8 *data;
+ guint size;
+ GstSDPMessage *sdp_msg;
+ const GstSDPMedia *sdp_media;
+ const GstSDPBandwidth *bw;
+ gint bandwidth_val = GPOINTER_TO_INT (user_data);
+
+ fail_unless (gst_rtsp_message_get_body (response, &data, &size)
+ == GST_RTSP_OK);
+ gst_sdp_message_new (&sdp_msg);
+ fail_unless (gst_sdp_message_parse_buffer (data, size, sdp_msg)
+ == GST_SDP_OK);
+
+ /* session description */
+ /* v= */
+ fail_unless (gst_sdp_message_get_version (sdp_msg) != NULL);
+ /* o= */
+ fail_unless (gst_sdp_message_get_origin (sdp_msg) != NULL);
+ /* s= */
+ fail_unless (gst_sdp_message_get_session_name (sdp_msg) != NULL);
+ /* t=0 0 */
+ fail_unless (gst_sdp_message_times_len (sdp_msg) == 0);
+
+ /* verify number of medias */
+ fail_unless (gst_sdp_message_medias_len (sdp_msg) == 1);
+
+ /* media description */
+ sdp_media = gst_sdp_message_get_media (sdp_msg, 0);
+ fail_unless (sdp_media != NULL);
+
+ /* m= */
+ fail_unless (gst_sdp_media_get_media (sdp_media) != NULL);
+
+ /* media bandwidth */
+ if (bandwidth_val) {
+ fail_unless (gst_sdp_media_bandwidths_len (sdp_media) == 1);
+ bw = gst_sdp_media_get_bandwidth (sdp_media, 0);
+ fail_unless (bw != NULL);
+ fail_unless (g_strcmp0 (bw->bwtype, "AS") == 0);
+ fail_unless (bw->bandwidth == bandwidth_val);
+ } else {
+ fail_unless (gst_sdp_media_bandwidths_len (sdp_media) == 0);
+ }
+
+ gst_sdp_message_free (sdp_msg);
+
+ return TRUE;
+}
+
+static void
+test_client_sdp (const gchar * launch_line, guint * bandwidth_val)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ /* simple DESCRIBE for an existing url */
+ client = setup_client (launch_line);
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+
+ gst_rtsp_client_set_send_func (client, test_response_sdp,
+ (gpointer) bandwidth_val, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ teardown_client (client);
+}
+
+GST_START_TEST (test_client_sdp_with_max_bitrate_tag)
+{
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"maximum-bitrate=(uint)50000000\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (50000));
+
+
+ /* max-bitrate=0: no bandwidth line */
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"maximum-bitrate=(uint)0\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (0));
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_sdp_with_bitrate_tag)
+{
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"bitrate=(uint)7000000\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (7000));
+
+ /* bitrate=0: no bandwdith line */
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"bitrate=(uint)0\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (0));
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_sdp_with_max_bitrate_and_bitrate_tags)
+{
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"bitrate=(uint)7000000,maximum-bitrate=(uint)50000000\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (50000));
+
+ /* max-bitrate is zero: fallback to bitrate */
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"bitrate=(uint)7000000,maximum-bitrate=(uint)0\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (7000));
+
+ /* max-bitrate=bitrate=0o: no bandwidth line */
+ test_client_sdp ("videotestsrc "
+ "! taginject tags=\"bitrate=(uint)0,maximum-bitrate=(uint)0\" "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+ GUINT_TO_POINTER (0));
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_sdp_with_no_bitrate_tags)
+{
+ test_client_sdp ("videotestsrc "
+ "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", NULL);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspclient_suite (void)
+{
+ Suite *s = suite_create ("rtspclient");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_request);
+ tcase_add_test (tc, test_options);
+ tcase_add_test (tc, test_describe);
+ tcase_add_test (tc, test_client_multicast_transport_404);
+ tcase_add_test (tc, test_client_multicast_transport);
+ tcase_add_test (tc, test_client_multicast_ignore_transport_specific);
+ tcase_add_test (tc, test_client_multicast_invalid_transport_specific);
+ tcase_add_test (tc, test_client_multicast_transport_specific);
+ tcase_add_test (tc, test_client_sdp_with_max_bitrate_tag);
+ tcase_add_test (tc, test_client_sdp_with_bitrate_tag);
+ tcase_add_test (tc, test_client_sdp_with_max_bitrate_and_bitrate_tags);
+ tcase_add_test (tc, test_client_sdp_with_no_bitrate_tags);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspclient);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-media-factory.h>
+
+GST_START_TEST (test_launch)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream;
+ GstRTSPTimeRange *range;
+ gchar *str;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream != NULL);
+
+ /* fails, need to be prepared */
+ str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ fail_unless (str == NULL);
+
+ fail_unless (gst_rtsp_range_parse ("npt=5.0-", &range) == GST_RTSP_OK);
+ /* fails, need to be prepared */
+ fail_if (gst_rtsp_media_seek (media, range));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ fail_unless (g_str_equal (str, "npt=0-"));
+ g_free (str);
+
+ str = gst_rtsp_media_get_range_string (media, TRUE, GST_RTSP_RANGE_NPT);
+ fail_unless (g_str_equal (str, "npt=0-"));
+ g_free (str);
+
+ fail_unless (gst_rtsp_media_seek (media, range));
+
+ str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ fail_unless (g_str_equal (str, "npt=5-"));
+ g_free (str);
+
+ str = gst_rtsp_media_get_range_string (media, TRUE, GST_RTSP_RANGE_NPT);
+ fail_unless (g_str_equal (str, "npt=5-"));
+ g_free (str);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ /* should fail again */
+ str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+ fail_unless (str == NULL);
+ fail_if (gst_rtsp_media_seek (media, range));
+
+ gst_rtsp_range_free (range);
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media)
+{
+ GstRTSPMedia *media;
+ GstElement *bin, *e1, *e2;
+
+ bin = gst_bin_new ("bin");
+ fail_if (bin == NULL);
+
+ e1 = gst_element_factory_make ("videotestsrc", NULL);
+ fail_if (e1 == NULL);
+
+ e2 = gst_element_factory_make ("rtpvrawpay", "pay0");
+ fail_if (e2 == NULL);
+ g_object_set (e2, "pt", 96, NULL);
+
+ gst_bin_add_many (GST_BIN_CAST (bin), e1, e2, NULL);
+ gst_element_link_many (e1, e2, NULL);
+
+ media = gst_rtsp_media_new (bin);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ g_object_unref (media);
+}
+
+GST_END_TEST;
+
+static void
+test_prepare_reusable (GstRTSPThreadPool * pool, const gchar * launch_line)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPThread *thread;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory, launch_line);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ g_object_set (G_OBJECT (media), "reusable", TRUE, NULL);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (gst_rtsp_media_unprepare (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ g_object_unref (media);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_START_TEST (test_media_prepare)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ pool = gst_rtsp_thread_pool_new ();
+
+ /* test non-reusable media first */
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (gst_rtsp_media_unprepare (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_if (gst_rtsp_media_prepare (media, thread));
+
+ g_object_unref (media);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+
+ /* test reusable media */
+ test_prepare_reusable (pool, "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+ test_prepare_reusable (pool,
+ "( videotestsrc is-live=true ! rtpvrawpay pt=96 name=pay0 )");
+
+ g_object_unref (pool);
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static void
+on_notify_caps (GstPad * pad, GParamSpec * pspec, GstElement * pay)
+{
+ GstCaps *caps;
+ static gboolean have_caps = FALSE;
+
+ g_object_get (pad, "caps", &caps, NULL);
+
+ GST_DEBUG ("notify %" GST_PTR_FORMAT, caps);
+
+ if (caps) {
+ if (!have_caps) {
+ g_signal_emit_by_name (pay, "pad-added", pad);
+ g_signal_emit_by_name (pay, "no-more-pads", NULL);
+ have_caps = TRUE;
+ }
+ gst_caps_unref (caps);
+ } else {
+ if (have_caps) {
+ g_signal_emit_by_name (pay, "pad-removed", pad);
+ have_caps = FALSE;
+ }
+ }
+}
+
+GST_START_TEST (test_media_dyn_prepare)
+{
+ GstRTSPMedia *media;
+ GstElement *bin, *src, *pay;
+ GstElement *pipeline;
+ GstPad *srcpad;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ bin = gst_bin_new ("bin");
+ fail_if (bin == NULL);
+
+ src = gst_element_factory_make ("videotestsrc", NULL);
+ fail_if (src == NULL);
+
+ pay = gst_element_factory_make ("rtpvrawpay", "dynpay0");
+ fail_if (pay == NULL);
+ g_object_set (pay, "pt", 96, NULL);
+
+ gst_bin_add_many (GST_BIN_CAST (bin), src, pay, NULL);
+ gst_element_link_many (src, pay, NULL);
+
+ media = gst_rtsp_media_new (bin);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ g_object_set (G_OBJECT (media), "reusable", TRUE, NULL);
+
+ pipeline = gst_pipeline_new ("media-pipeline");
+ gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+ gst_rtsp_media_collect_streams (media);
+
+ srcpad = gst_element_get_static_pad (pay, "src");
+
+ g_signal_connect (srcpad, "notify::caps", (GCallback) on_notify_caps, pay);
+
+ pool = gst_rtsp_thread_pool_new ();
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+ fail_unless (gst_rtsp_media_unprepare (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+ fail_unless (gst_rtsp_media_unprepare (media));
+ fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+ gst_object_unref (srcpad);
+ g_object_unref (media);
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_prepare_port_alloc_fail)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPAddressPool *addrpool;
+
+ pool = gst_rtsp_thread_pool_new ();
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( fakesrc is-live=true ! text/plain ! rtpgstpay name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ addrpool = gst_rtsp_address_pool_new ();
+ fail_unless (gst_rtsp_address_pool_add_range (addrpool, "192.168.1.1",
+ "192.168.1.1", 6000, 6001, 0));
+ gst_rtsp_media_set_address_pool (media, addrpool);
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_if (gst_rtsp_media_prepare (media, thread));
+
+ g_object_unref (media);
+ g_object_unref (addrpool);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_take_pipeline)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstElement *pipeline;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+ gst_rtsp_media_factory_set_launch (factory,
+ "( fakesrc ! text/plain ! rtpgstpay name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ pipeline = gst_pipeline_new ("media-pipeline");
+ gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+ g_object_unref (media);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_reset)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ pool = gst_rtsp_thread_pool_new ();
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ gst_rtsp_url_parse ("rtsp://localhost:8554/test", &url);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (gst_rtsp_media_suspend (media));
+ fail_unless (gst_rtsp_media_unprepare (media));
+ g_object_unref (media);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+ gst_rtsp_media_set_suspend_mode (media, GST_RTSP_SUSPEND_MODE_RESET);
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+ fail_unless (gst_rtsp_media_suspend (media));
+ fail_unless (gst_rtsp_media_unprepare (media));
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspmedia_suite (void)
+{
+ Suite *s = suite_create ("rtspmedia");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_launch);
+ tcase_add_test (tc, test_media);
+ tcase_add_test (tc, test_media_prepare);
+ tcase_add_test (tc, test_media_dyn_prepare);
+ tcase_add_test (tc, test_media_prepare_port_alloc_fail);
+ tcase_add_test (tc, test_media_take_pipeline);
+ tcase_add_test (tc, test_media_reset);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspmedia);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-media-factory.h>
+
+GST_START_TEST (test_parse_error)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+
+ gst_rtsp_media_factory_set_launch (factory, "foo");
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+ ASSERT_CRITICAL (gst_rtsp_media_factory_create_element (factory, url));
+ ASSERT_CRITICAL (gst_rtsp_media_factory_construct (factory, url));
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_launch)
+{
+ GstRTSPMediaFactory *factory;
+ GstElement *element;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ fail_unless (GST_IS_BIN (element));
+ fail_if (GST_IS_PIPELINE (element));
+ gst_object_unref (element);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_launch_construct)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media, *media2;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ media2 = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media2));
+ fail_if (media == media2);
+
+ g_object_unref (media);
+ g_object_unref (media2);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_shared)
+{
+ GstRTSPMediaFactory *factory;
+ GstElement *element;
+ GstRTSPMedia *media, *media2;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ fail_unless (gst_rtsp_media_factory_is_shared (factory));
+
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ fail_unless (GST_IS_BIN (element));
+ fail_if (GST_IS_PIPELINE (element));
+ gst_object_unref (element);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ media2 = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media2));
+ fail_unless (media == media2);
+
+ g_object_unref (media);
+ g_object_unref (media2);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_addresspool)
+{
+ GstRTSPMediaFactory *factory;
+ GstElement *element;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPAddressPool *pool, *tmppool;
+ GstRTSPStream *stream;
+ GstRTSPAddress *addr;
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 "
+ " audiotestsrc ! audioconvert ! rtpL16pay name=pay1 )");
+
+ pool = gst_rtsp_address_pool_new ();
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.1", "233.252.0.1", 5000, 5001, 3));
+
+ gst_rtsp_media_factory_set_address_pool (factory, pool);
+
+ tmppool = gst_rtsp_media_factory_get_address_pool (factory);
+ fail_unless (pool == tmppool);
+ g_object_unref (tmppool);
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ fail_unless (GST_IS_BIN (element));
+ fail_if (GST_IS_PIPELINE (element));
+ gst_object_unref (element);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ tmppool = gst_rtsp_media_get_address_pool (media);
+ fail_unless (pool == tmppool);
+ g_object_unref (tmppool);
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream != NULL);
+
+ tmppool = gst_rtsp_stream_get_address_pool (stream);
+ fail_unless (pool == tmppool);
+ g_object_unref (tmppool);
+
+ addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+ fail_unless (addr != NULL);
+ fail_unless (addr->port == 5000);
+ fail_unless (addr->n_ports == 2);
+ fail_unless (addr->ttl == 3);
+ gst_rtsp_address_free (addr);
+
+ stream = gst_rtsp_media_get_stream (media, 1);
+ fail_unless (stream != NULL);
+
+ tmppool = gst_rtsp_stream_get_address_pool (stream);
+ fail_unless (pool == tmppool);
+ g_object_unref (tmppool);
+
+ addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+ fail_unless (addr == NULL);
+
+
+ g_object_unref (media);
+
+ g_object_unref (pool);
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_permissions)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPPermissions *perms;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ gst_rtsp_media_factory_add_role (factory, "admin",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ perms = gst_rtsp_media_factory_get_permissions (factory);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.access"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.construct"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
+ "media.factory.access"));
+ gst_rtsp_permissions_unref (perms);
+
+ perms = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (perms, "user",
+ "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+ "media.factory.construct", G_TYPE_BOOLEAN, FALSE, NULL);
+ gst_rtsp_media_factory_set_permissions (factory, perms);
+ gst_rtsp_permissions_unref (perms);
+
+ perms = gst_rtsp_media_factory_get_permissions (factory);
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.access"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.construct"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user",
+ "media.factory.access"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user",
+ "media.factory.construct"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
+ "media.factory.access"));
+ gst_rtsp_permissions_unref (perms);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ perms = gst_rtsp_media_get_permissions (media);
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.access"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+ "media.factory.construct"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user",
+ "media.factory.access"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user",
+ "media.factory.construct"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
+ "media.factory.access"));
+ gst_rtsp_permissions_unref (perms);
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_reset)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ gst_rtsp_url_parse ("rtsp://localhost:8554/test", &url);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ fail_if (gst_rtsp_media_get_suspend_mode (media) !=
+ GST_RTSP_SUSPEND_MODE_NONE);
+ g_object_unref (media);
+
+ gst_rtsp_media_factory_set_suspend_mode (factory,
+ GST_RTSP_SUSPEND_MODE_RESET);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+ fail_if (gst_rtsp_media_get_suspend_mode (media) !=
+ GST_RTSP_SUSPEND_MODE_RESET);
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspmediafactory_suite (void)
+{
+ Suite *s = suite_create ("rtspmediafactory");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_parse_error);
+ tcase_add_test (tc, test_launch);
+ tcase_add_test (tc, test_launch_construct);
+ tcase_add_test (tc, test_shared);
+ tcase_add_test (tc, test_addresspool);
+ tcase_add_test (tc, test_permissions);
+ tcase_add_test (tc, test_reset);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspmediafactory);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-mount-points.h>
+
+GST_START_TEST (test_create)
+{
+ GstRTSPMountPoints *mounts;
+ GstRTSPUrl *url, *url2;
+ GstRTSPMediaFactory *factory;
+
+ mounts = gst_rtsp_mount_points_new ();
+
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test2",
+ &url2) == GST_RTSP_OK);
+
+ fail_unless (gst_rtsp_mount_points_match (mounts, url->abspath,
+ NULL) == NULL);
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+ fail_unless (gst_rtsp_mount_points_match (mounts, url->abspath,
+ NULL) == factory);
+ g_object_unref (factory);
+ fail_unless (gst_rtsp_mount_points_match (mounts, url2->abspath,
+ NULL) == NULL);
+
+ gst_rtsp_mount_points_remove_factory (mounts, "/test");
+
+ fail_unless (gst_rtsp_mount_points_match (mounts, url->abspath,
+ NULL) == NULL);
+ fail_unless (gst_rtsp_mount_points_match (mounts, url2->abspath,
+ NULL) == NULL);
+
+ gst_rtsp_url_free (url);
+ gst_rtsp_url_free (url2);
+
+ g_object_unref (mounts);
+}
+
+GST_END_TEST;
+
+static const gchar *paths[] = {
+ "/test",
+ "/booz/fooz",
+ "/booz/foo/zoop",
+ "/tark/bar",
+ "/tark/bar/baz",
+ "/tark/bar/baz/t",
+ "/boozop",
+};
+
+GST_START_TEST (test_match)
+{
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *f[G_N_ELEMENTS (paths)], *tmp;
+ gint i, matched;
+
+ mounts = gst_rtsp_mount_points_new ();
+
+ for (i = 0; i < G_N_ELEMENTS (paths); i++) {
+ f[i] = gst_rtsp_media_factory_new ();
+ gst_rtsp_mount_points_add_factory (mounts, paths[i], f[i]);
+ }
+
+ tmp = gst_rtsp_mount_points_match (mounts, "/test", &matched);
+ fail_unless (tmp == f[0]);
+ fail_unless (matched == 5);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/test/stream=1", &matched);
+ fail_unless (tmp == f[0]);
+ fail_unless (matched == 5);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/booz", &matched);
+ fail_unless (tmp == NULL);
+ tmp = gst_rtsp_mount_points_match (mounts, "/booz/foo", &matched);
+ fail_unless (tmp == NULL);
+ tmp = gst_rtsp_mount_points_match (mounts, "/booz/fooz", &matched);
+ fail_unless (tmp == f[1]);
+ fail_unless (matched == 10);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/booz/fooz/zoo", &matched);
+ fail_unless (tmp == f[1]);
+ fail_unless (matched == 10);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/booz/foo/zoop", &matched);
+ fail_unless (tmp == f[2]);
+ fail_unless (matched == 14);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar", &matched);
+ fail_unless (tmp == f[3]);
+ fail_unless (matched == 9);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar/boo", &matched);
+ fail_unless (tmp == f[3]);
+ fail_unless (matched == 9);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar/ba", &matched);
+ fail_unless (tmp == f[3]);
+ fail_unless (matched == 9);
+ g_object_unref (tmp);
+ tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar/baz", &matched);
+ fail_unless (tmp == f[4]);
+ fail_unless (matched == 13);
+ g_object_unref (tmp);
+
+ g_object_unref (mounts);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspmountpoints_suite (void)
+{
+ Suite *s = suite_create ("rtspmountpoints");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_create);
+ tcase_add_test (tc, test_match);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspmountpoints);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Sebastian Rasmussen <sebras@hotmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-permissions.h>
+
+GST_START_TEST (test_permissions)
+{
+ GstRTSPPermissions *perms;
+ GstRTSPPermissions *copy;
+
+ perms = gst_rtsp_permissions_new ();
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "missing", "permission1"));
+ gst_rtsp_permissions_unref (perms);
+
+ perms = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (perms, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "missing"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "missing", "permission1"));
+ copy = GST_RTSP_PERMISSIONS (gst_mini_object_copy (GST_MINI_OBJECT (perms)));
+ gst_rtsp_permissions_unref (perms);
+ fail_unless (gst_rtsp_permissions_is_allowed (copy, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (copy, "user", "permission2"));
+ gst_rtsp_permissions_unref (copy);
+
+ perms = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (perms, "admin",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_permissions_add_role (perms, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission2"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ gst_rtsp_permissions_unref (perms);
+
+ perms = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (perms, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ gst_rtsp_permissions_add_role (perms, "user",
+ "permission1", G_TYPE_BOOLEAN, FALSE,
+ "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ gst_rtsp_permissions_unref (perms);
+
+ perms = gst_rtsp_permissions_new ();
+ gst_rtsp_permissions_add_role (perms, "admin",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_rtsp_permissions_add_role (perms, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission2"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ gst_rtsp_permissions_remove_role (perms, "user");
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+ fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission2"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+ fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+ gst_rtsp_permissions_unref (perms);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtsppermissions_suite (void)
+{
+ Suite *s = suite_create ("rtsppermissions");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_permissions);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtsppermissions);
--- /dev/null
+/* GStreamer
+ *
+ * unit test for GstRTSPServer
+ *
+ * Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
+ * @author David Svensson Fors <davidsf at axis dot com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/sdp/gstsdpmessage.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+
+#include <stdio.h>
+#include <netinet/in.h>
+
+#include "rtsp-server.h"
+
+#define VIDEO_PIPELINE "videotestsrc ! " \
+ "video/x-raw,width=352,height=288 ! " \
+ "rtpgstpay name=pay0 pt=96"
+#define AUDIO_PIPELINE "audiotestsrc ! " \
+ "audio/x-raw,rate=8000 ! " \
+ "rtpgstpay name=pay1 pt=97"
+
+#define TEST_MOUNT_POINT "/test"
+#define TEST_PROTO "RTP/AVP"
+#define TEST_ENCODING "X-GST"
+#define TEST_CLOCK_RATE "90000"
+
+/* tested rtsp server */
+static GstRTSPServer *server = NULL;
+
+/* tcp port that the test server listens for rtsp requests on */
+static gint test_port = 0;
+
+/* id of the server's source within the GMainContext */
+static guint source_id;
+
+/* iterate the default main loop until there are no events to dispatch */
+static void
+iterate (void)
+{
+ while (g_main_context_iteration (NULL, FALSE)) {
+ GST_DEBUG ("iteration");
+ }
+}
+
+static void
+get_client_ports_full (GstRTSPRange * range, GSocket ** rtp_socket,
+ GSocket ** rtcp_socket)
+{
+ GSocket *rtp = NULL;
+ GSocket *rtcp = NULL;
+ gint rtp_port = 0;
+ gint rtcp_port;
+ GInetAddress *anyaddr = g_inet_address_new_any (G_SOCKET_FAMILY_IPV4);
+ GSocketAddress *sockaddr;
+ gboolean bound;
+
+ for (;;) {
+ if (rtp_port != 0)
+ rtp_port += 2;
+
+ rtp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ fail_unless (rtp != NULL);
+
+ sockaddr = g_inet_socket_address_new (anyaddr, rtp_port);
+ fail_unless (sockaddr != NULL);
+ bound = g_socket_bind (rtp, sockaddr, FALSE, NULL);
+ g_object_unref (sockaddr);
+ if (!bound) {
+ g_object_unref (rtp);
+ continue;
+ }
+
+ sockaddr = g_socket_get_local_address (rtp, NULL);
+ fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
+ rtp_port =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
+ g_object_unref (sockaddr);
+
+ if (rtp_port % 2 != 0) {
+ rtp_port += 1;
+ g_object_unref (rtp);
+ continue;
+ }
+
+ rtcp_port = rtp_port + 1;
+
+ rtcp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ fail_unless (rtcp != NULL);
+
+ sockaddr = g_inet_socket_address_new (anyaddr, rtcp_port);
+ fail_unless (sockaddr != NULL);
+ bound = g_socket_bind (rtcp, sockaddr, FALSE, NULL);
+ g_object_unref (sockaddr);
+ if (!bound) {
+ g_object_unref (rtp);
+ g_object_unref (rtcp);
+ continue;
+ }
+
+ sockaddr = g_socket_get_local_address (rtcp, NULL);
+ fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
+ fail_unless (rtcp_port ==
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr)));
+ g_object_unref (sockaddr);
+
+ break;
+ }
+
+ range->min = rtp_port;
+ range->max = rtcp_port;
+ if (rtp_socket)
+ *rtp_socket = rtp;
+ else
+ g_object_unref (rtp);
+ if (rtcp_socket)
+ *rtcp_socket = rtcp;
+ else
+ g_object_unref (rtcp);
+ GST_DEBUG ("client_port=%d-%d", range->min, range->max);
+ g_object_unref (anyaddr);
+}
+
+/* get a free rtp/rtcp client port pair */
+static void
+get_client_ports (GstRTSPRange * range)
+{
+ get_client_ports_full (range, NULL, NULL);
+}
+
+/* start the tested rtsp server */
+static void
+start_server (void)
+{
+ GstRTSPMountPoints *mounts;
+ gchar *service;
+ GstRTSPMediaFactory *factory;
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = gst_rtsp_media_factory_new ();
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+ g_object_unref (mounts);
+
+ /* set port to any */
+ gst_rtsp_server_set_service (server, "0");
+
+ /* attach to default main context */
+ source_id = gst_rtsp_server_attach (server, NULL);
+ fail_if (source_id == 0);
+
+ /* get port */
+ service = gst_rtsp_server_get_service (server);
+ test_port = atoi (service);
+ fail_unless (test_port != 0);
+ g_free (service);
+
+ GST_DEBUG ("rtsp server listening on port %d", test_port);
+}
+
+/* stop the tested rtsp server */
+static void
+stop_server (void)
+{
+ g_source_remove (source_id);
+ source_id = 0;
+
+ GST_DEBUG ("rtsp server stopped");
+}
+
+/* create an rtsp connection to the server on test_port */
+static GstRTSPConnection *
+connect_to_server (gint port, const gchar * mount_point)
+{
+ GstRTSPConnection *conn = NULL;
+ gchar *address;
+ gchar *uri_string;
+ GstRTSPUrl *url = NULL;
+
+ address = gst_rtsp_server_get_address (server);
+ uri_string = g_strdup_printf ("rtsp://%s:%d%s", address, port, mount_point);
+ g_free (address);
+ fail_unless (gst_rtsp_url_parse (uri_string, &url) == GST_RTSP_OK);
+ g_free (uri_string);
+
+ fail_unless (gst_rtsp_connection_create (url, &conn) == GST_RTSP_OK);
+ gst_rtsp_url_free (url);
+
+ fail_unless (gst_rtsp_connection_connect (conn, NULL) == GST_RTSP_OK);
+
+ return conn;
+}
+
+/* create an rtsp request */
+static GstRTSPMessage *
+create_request (GstRTSPConnection * conn, GstRTSPMethod method,
+ const gchar * control)
+{
+ GstRTSPMessage *request = NULL;
+ gchar *base_uri;
+ gchar *full_uri;
+
+ base_uri = gst_rtsp_url_get_request_uri (gst_rtsp_connection_get_url (conn));
+ full_uri = g_strdup_printf ("%s/%s", base_uri, control ? control : "");
+ g_free (base_uri);
+ if (gst_rtsp_message_new_request (&request, method, full_uri) != GST_RTSP_OK) {
+ GST_DEBUG ("failed to create request object");
+ g_free (full_uri);
+ return NULL;
+ }
+ g_free (full_uri);
+ return request;
+}
+
+/* send an rtsp request */
+static gboolean
+send_request (GstRTSPConnection * conn, GstRTSPMessage * request)
+{
+ if (gst_rtsp_connection_send (conn, request, NULL) != GST_RTSP_OK) {
+ GST_DEBUG ("failed to send request");
+ return FALSE;
+ }
+ return TRUE;
+}
+
+/* read rtsp response. response must be freed by the caller */
+static GstRTSPMessage *
+read_response (GstRTSPConnection * conn)
+{
+ GstRTSPMessage *response = NULL;
+
+ if (gst_rtsp_message_new (&response) != GST_RTSP_OK) {
+ GST_DEBUG ("failed to create response object");
+ return NULL;
+ }
+ if (gst_rtsp_connection_receive (conn, response, NULL) != GST_RTSP_OK) {
+ GST_DEBUG ("failed to read response");
+ gst_rtsp_message_free (response);
+ return NULL;
+ }
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+ return response;
+}
+
+/* send an rtsp request and receive response. gchar** parameters are out
+ * parameters that have to be freed by the caller */
+static GstRTSPStatusCode
+do_request_full (GstRTSPConnection * conn, GstRTSPMethod method,
+ const gchar * control, const gchar * session_in, const gchar * transport_in,
+ const gchar * range_in, const gchar * require_in,
+ gchar ** content_type, gchar ** content_base, gchar ** body,
+ gchar ** session_out, gchar ** transport_out, gchar ** range_out,
+ gchar ** unsupported_out)
+{
+ GstRTSPMessage *request;
+ GstRTSPMessage *response;
+ GstRTSPStatusCode code;
+ gchar *value;
+
+ /* create request */
+ request = create_request (conn, method, control);
+
+ /* add headers */
+ if (session_in) {
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session_in);
+ }
+ if (transport_in) {
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_TRANSPORT, transport_in);
+ }
+ if (range_in) {
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_RANGE, range_in);
+ }
+ if (require_in) {
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_REQUIRE, require_in);
+ }
+
+ /* send request */
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+
+ iterate ();
+
+ /* read response */
+ response = read_response (conn);
+
+ /* check status line */
+ gst_rtsp_message_parse_response (response, &code, NULL, NULL);
+ if (code != GST_RTSP_STS_OK) {
+ if (unsupported_out != NULL && code == GST_RTSP_STS_OPTION_NOT_SUPPORTED) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_UNSUPPORTED,
+ &value, 0);
+ *unsupported_out = g_strdup (value);
+ }
+ gst_rtsp_message_free (response);
+ return code;
+ }
+
+ /* get information from response */
+ if (content_type) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_TYPE,
+ &value, 0);
+ *content_type = g_strdup (value);
+ }
+ if (content_base) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
+ &value, 0);
+ *content_base = g_strdup (value);
+ }
+ if (body) {
+ *body = g_malloc (response->body_size + 1);
+ strncpy (*body, (gchar *) response->body, response->body_size);
+ }
+ if (session_out) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &value, 0);
+
+ value = g_strdup (value);
+
+ /* Remove the timeout */
+ if (value) {
+ char *pos = strchr (value, ';');
+ if (pos)
+ *pos = 0;
+ }
+ if (session_in) {
+ /* check that we got the same session back */
+ fail_unless (!g_strcmp0 (value, session_in));
+ }
+ *session_out = value;
+ }
+ if (transport_out) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT, &value, 0);
+ *transport_out = g_strdup (value);
+ }
+ if (range_out) {
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_RANGE, &value, 0);
+ *range_out = g_strdup (value);
+ }
+
+ gst_rtsp_message_free (response);
+ return code;
+}
+
+/* send an rtsp request and receive response. gchar** parameters are out
+ * parameters that have to be freed by the caller */
+static GstRTSPStatusCode
+do_request (GstRTSPConnection * conn, GstRTSPMethod method,
+ const gchar * control, const gchar * session_in,
+ const gchar * transport_in, const gchar * range_in,
+ gchar ** content_type, gchar ** content_base, gchar ** body,
+ gchar ** session_out, gchar ** transport_out, gchar ** range_out)
+{
+ return do_request_full (conn, method, control, session_in, transport_in,
+ range_in, NULL, content_type, content_base, body, session_out,
+ transport_out, range_out, NULL);
+}
+
+/* send an rtsp request with a method and a session, and receive response */
+static GstRTSPStatusCode
+do_simple_request (GstRTSPConnection * conn, GstRTSPMethod method,
+ const gchar * session)
+{
+ return do_request (conn, method, NULL, session, NULL, NULL, NULL,
+ NULL, NULL, NULL, NULL, NULL);
+}
+
+/* send a DESCRIBE request and receive response. returns a received
+ * GstSDPMessage that must be freed by the caller */
+static GstSDPMessage *
+do_describe (GstRTSPConnection * conn, const gchar * mount_point)
+{
+ GstSDPMessage *sdp_message;
+ gchar *content_type;
+ gchar *content_base;
+ gchar *body;
+ gchar *address;
+ gchar *expected_content_base;
+
+ /* send DESCRIBE request */
+ fail_unless (do_request (conn, GST_RTSP_DESCRIBE, NULL, NULL, NULL, NULL,
+ &content_type, &content_base, &body, NULL, NULL, NULL) ==
+ GST_RTSP_STS_OK);
+
+ /* check response values */
+ fail_unless (!g_strcmp0 (content_type, "application/sdp"));
+ address = gst_rtsp_server_get_address (server);
+ expected_content_base =
+ g_strdup_printf ("rtsp://%s:%d%s/", address, test_port, mount_point);
+ fail_unless (!g_strcmp0 (content_base, expected_content_base));
+
+ /* create sdp message */
+ fail_unless (gst_sdp_message_new (&sdp_message) == GST_SDP_OK);
+ fail_unless (gst_sdp_message_parse_buffer ((guint8 *) body,
+ strlen (body), sdp_message) == GST_SDP_OK);
+
+ /* clean up */
+ g_free (content_type);
+ g_free (content_base);
+ g_free (body);
+ g_free (address);
+ g_free (expected_content_base);
+
+ return sdp_message;
+}
+
+/* send a SETUP request and receive response. if *session is not NULL,
+ * it is used in the request. otherwise, *session is set to a returned
+ * session string that must be freed by the caller. the returned
+ * transport must be freed by the caller. */
+static GstRTSPStatusCode
+do_setup_full (GstRTSPConnection * conn, const gchar * control,
+ const GstRTSPRange * client_ports, const gchar * require, gchar ** session,
+ GstRTSPTransport ** transport, gchar ** unsupported)
+{
+ GstRTSPStatusCode code;
+ gchar *session_in = NULL;
+ gchar *transport_string_in = NULL;
+ gchar **session_out = NULL;
+ gchar *transport_string_out = NULL;
+
+ /* prepare and send SETUP request */
+ if (session) {
+ if (*session) {
+ session_in = *session;
+ } else {
+ session_out = session;
+ }
+ }
+ transport_string_in =
+ g_strdup_printf (TEST_PROTO ";unicast;client_port=%d-%d",
+ client_ports->min, client_ports->max);
+ code =
+ do_request_full (conn, GST_RTSP_SETUP, control, session_in,
+ transport_string_in, NULL, require, NULL, NULL, NULL, session_out,
+ &transport_string_out, NULL, unsupported);
+ g_free (transport_string_in);
+
+ if (transport_string_out) {
+ /* create transport */
+ fail_unless (gst_rtsp_transport_new (transport) == GST_RTSP_OK);
+ fail_unless (gst_rtsp_transport_parse (transport_string_out,
+ *transport) == GST_RTSP_OK);
+ g_free (transport_string_out);
+ }
+ GST_INFO ("code=%d", code);
+ return code;
+}
+
+/* send a SETUP request and receive response. if *session is not NULL,
+ * it is used in the request. otherwise, *session is set to a returned
+ * session string that must be freed by the caller. the returned
+ * transport must be freed by the caller. */
+static GstRTSPStatusCode
+do_setup (GstRTSPConnection * conn, const gchar * control,
+ const GstRTSPRange * client_ports, gchar ** session,
+ GstRTSPTransport ** transport)
+{
+ return do_setup_full (conn, control, client_ports, NULL, session, transport,
+ NULL);
+}
+
+/* fixture setup function */
+static void
+setup (void)
+{
+ server = gst_rtsp_server_new ();
+}
+
+/* fixture clean-up function */
+static void
+teardown (void)
+{
+ if (server) {
+ g_object_unref (server);
+ server = NULL;
+ }
+ test_port = 0;
+}
+
+GST_START_TEST (test_connect)
+{
+ GstRTSPConnection *conn;
+
+ start_server ();
+
+ /* connect to server */
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* clean up */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+
+ /* iterate so the clean-up can finish */
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_describe)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ gint32 format;
+ gchar *expected_rtpmap;
+ const gchar *rtpmap;
+ const gchar *control_video;
+ const gchar *control_audio;
+
+ start_server ();
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* send DESCRIBE request */
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+
+ /* check video sdp */
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ fail_unless (!g_strcmp0 (gst_sdp_media_get_proto (sdp_media), TEST_PROTO));
+ fail_unless (gst_sdp_media_formats_len (sdp_media) == 1);
+ sscanf (gst_sdp_media_get_format (sdp_media, 0), "%" G_GINT32_FORMAT,
+ &format);
+ expected_rtpmap =
+ g_strdup_printf ("%d " TEST_ENCODING "/" TEST_CLOCK_RATE, format);
+ rtpmap = gst_sdp_media_get_attribute_val (sdp_media, "rtpmap");
+ fail_unless (!g_strcmp0 (rtpmap, expected_rtpmap));
+ g_free (expected_rtpmap);
+ control_video = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ fail_unless (!g_strcmp0 (control_video, "stream=0"));
+
+ /* check audio sdp */
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ fail_unless (!g_strcmp0 (gst_sdp_media_get_proto (sdp_media), TEST_PROTO));
+ fail_unless (gst_sdp_media_formats_len (sdp_media) == 1);
+ sscanf (gst_sdp_media_get_format (sdp_media, 0), "%" G_GINT32_FORMAT,
+ &format);
+ expected_rtpmap =
+ g_strdup_printf ("%d " TEST_ENCODING "/" TEST_CLOCK_RATE, format);
+ rtpmap = gst_sdp_media_get_attribute_val (sdp_media, "rtpmap");
+ fail_unless (!g_strcmp0 (rtpmap, expected_rtpmap));
+ g_free (expected_rtpmap);
+ control_audio = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ fail_unless (!g_strcmp0 (control_audio, "stream=1"));
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_describe_non_existing_mount_point)
+{
+ GstRTSPConnection *conn;
+
+ start_server ();
+
+ /* send DESCRIBE request for a non-existing mount point
+ * and check that we get a 404 Not Found */
+ conn = connect_to_server (test_port, "/non-existing");
+ fail_unless (do_simple_request (conn, GST_RTSP_DESCRIBE, NULL)
+ == GST_RTSP_STS_NOT_FOUND);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_ports;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+
+ start_server ();
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_ports);
+
+ /* send SETUP request for video */
+ fail_unless (do_setup (conn, video_control, &client_ports, &session,
+ &video_transport) == GST_RTSP_STS_OK);
+ GST_DEBUG ("set up video %s, got session '%s'", video_control, session);
+
+ /* check response from SETUP */
+ fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
+ fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
+ fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
+ fail_unless (video_transport->mode_play);
+ gst_rtsp_transport_free (video_transport);
+
+ /* send SETUP request for audio */
+ fail_unless (do_setup (conn, audio_control, &client_ports, &session,
+ &audio_transport) == GST_RTSP_STS_OK);
+ GST_DEBUG ("set up audio %s with session '%s'", audio_control, session);
+
+ /* check response from SETUP */
+ fail_unless (audio_transport->trans == GST_RTSP_TRANS_RTP);
+ fail_unless (audio_transport->profile == GST_RTSP_PROFILE_AVP);
+ fail_unless (audio_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
+ fail_unless (audio_transport->mode_play);
+ gst_rtsp_transport_free (audio_transport);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_free (session);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_with_require_header)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ GstRTSPRange client_ports;
+ gchar *session = NULL;
+ gchar *unsupported = NULL;
+ GstRTSPTransport *video_transport = NULL;
+
+ start_server ();
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_ports);
+
+ /* send SETUP request for video, with single Require header */
+ fail_unless_equals_int (do_setup_full (conn, video_control, &client_ports,
+ "funky-feature", &session, &video_transport, &unsupported),
+ GST_RTSP_STS_OPTION_NOT_SUPPORTED);
+ fail_unless_equals_string (unsupported, "funky-feature");
+ g_free (unsupported);
+ unsupported = NULL;
+
+ /* send SETUP request for video, with multiple Require headers */
+ fail_unless_equals_int (do_setup_full (conn, video_control, &client_ports,
+ "funky-feature, foo-bar, superburst", &session, &video_transport,
+ &unsupported), GST_RTSP_STS_OPTION_NOT_SUPPORTED);
+ fail_unless_equals_string (unsupported, "funky-feature, foo-bar, superburst");
+ g_free (unsupported);
+ unsupported = NULL;
+
+ /* ok, just do a normal setup then (make sure that still works) */
+ fail_unless_equals_int (do_setup (conn, video_control, &client_ports,
+ &session, &video_transport), GST_RTSP_STS_OK);
+
+ GST_DEBUG ("set up video %s, got session '%s'", video_control, session);
+
+ /* check response from SETUP */
+ fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
+ fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
+ fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
+ fail_unless (video_transport->mode_play);
+ gst_rtsp_transport_free (video_transport);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_free (session);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_non_existing_stream)
+{
+ GstRTSPConnection *conn;
+ GstRTSPRange client_ports;
+
+ start_server ();
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ get_client_ports (&client_ports);
+
+ /* send SETUP request with a non-existing stream and check that we get a
+ * 404 Not Found */
+ fail_unless (do_setup (conn, "stream=7", &client_ports, NULL,
+ NULL) == GST_RTSP_STS_NOT_FOUND);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+static void
+receive_rtp (GSocket * socket, GSocketAddress ** addr)
+{
+ GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
+
+ for (;;) {
+ gssize bytes;
+ GstMapInfo map = GST_MAP_INFO_INIT;
+ GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
+
+ gst_buffer_map (buffer, &map, GST_MAP_WRITE);
+ bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
+ map.maxsize, NULL, NULL);
+ fail_unless (bytes > 0);
+ gst_buffer_unmap (buffer, &map);
+ gst_buffer_set_size (buffer, bytes);
+
+ if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer)) {
+ gst_rtp_buffer_unmap (&rtpbuffer);
+ break;
+ }
+
+ if (addr)
+ g_clear_object (addr);
+ }
+
+ gst_buffer_unref (buffer);
+}
+
+static void
+receive_rtcp (GSocket * socket, GSocketAddress ** addr, GstRTCPType type)
+{
+ GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
+
+ for (;;) {
+ gssize bytes;
+ GstMapInfo map = GST_MAP_INFO_INIT;
+
+ gst_buffer_map (buffer, &map, GST_MAP_WRITE);
+ bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
+ map.maxsize, NULL, NULL);
+ fail_unless (bytes > 0);
+ gst_buffer_unmap (buffer, &map);
+ gst_buffer_set_size (buffer, bytes);
+
+ if (gst_rtcp_buffer_validate (buffer)) {
+ GstRTCPBuffer rtcpbuffer = GST_RTCP_BUFFER_INIT;
+ GstRTCPPacket packet;
+
+ if (type) {
+ fail_unless (gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcpbuffer));
+ fail_unless (gst_rtcp_buffer_get_first_packet (&rtcpbuffer, &packet));
+ do {
+ if (gst_rtcp_packet_get_type (&packet) == type) {
+ gst_rtcp_buffer_unmap (&rtcpbuffer);
+ goto done;
+ }
+ } while (gst_rtcp_packet_move_to_next (&packet));
+ gst_rtcp_buffer_unmap (&rtcpbuffer);
+ } else {
+ break;
+ }
+ }
+
+ if (addr)
+ g_clear_object (addr);
+ }
+
+done:
+
+ gst_buffer_unref (buffer);
+}
+
+static void
+do_test_play (const gchar * range)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GSocket *rtp_socket, *rtcp_socket;
+ gchar *range_out = NULL;
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup (conn, video_control, &client_port, &session,
+ &video_transport) == GST_RTSP_STS_OK);
+ fail_unless (do_setup (conn, audio_control, &client_port, &session,
+ &audio_transport) == GST_RTSP_STS_OK);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_request (conn, GST_RTSP_PLAY, NULL, session, NULL, range,
+ NULL, NULL, NULL, NULL, NULL, &range_out) == GST_RTSP_STS_OK);
+ if (range)
+ fail_unless_equals_string (range, range_out);
+ g_free (range_out);
+
+ receive_rtp (rtp_socket, NULL);
+ receive_rtcp (rtcp_socket, NULL, 0);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* FIXME: The rtsp-server always disconnects the transport before
+ * sending the RTCP BYE
+ * receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
+ */
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (rtp_socket);
+ g_object_unref (rtcp_socket);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+}
+
+
+GST_START_TEST (test_play)
+{
+ start_server ();
+
+ do_test_play (NULL);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_play_without_session)
+{
+ GstRTSPConnection *conn;
+
+ start_server ();
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* send PLAY request without a session and check that we get a
+ * 454 Session Not Found */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ NULL) == GST_RTSP_STS_SESSION_NOT_FOUND);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_bind_already_in_use)
+{
+ GstRTSPServer *serv;
+ GSocketService *service;
+ GError *error = NULL;
+ guint16 port;
+ gchar *port_str;
+
+ serv = gst_rtsp_server_new ();
+ service = g_socket_service_new ();
+
+ /* bind service to port */
+ port =
+ g_socket_listener_add_any_inet_port (G_SOCKET_LISTENER (service), NULL,
+ &error);
+ g_assert_no_error (error);
+
+ port_str = g_strdup_printf ("%d\n", port);
+
+ /* try to bind server to the same port */
+ g_object_set (serv, "service", port_str, NULL);
+ g_free (port_str);
+
+ /* attach to default main context */
+ fail_unless (gst_rtsp_server_attach (serv, NULL) == 0);
+
+ /* cleanup */
+ g_object_unref (serv);
+ g_socket_service_stop (service);
+ g_object_unref (service);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_multithreaded)
+{
+ GstRTSPThreadPool *pool;
+
+ pool = gst_rtsp_server_get_thread_pool (server);
+ gst_rtsp_thread_pool_set_max_threads (pool, 2);
+ g_object_unref (pool);
+
+ start_server ();
+
+ do_test_play (NULL);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+enum
+{
+ BLOCK_ME,
+ BLOCKED,
+ UNBLOCK
+};
+
+
+static void
+media_constructed_block (GstRTSPMediaFactory * factory,
+ GstRTSPMedia * media, gpointer user_data)
+{
+ gint *block_state = user_data;
+
+ g_mutex_lock (&check_mutex);
+
+ *block_state = BLOCKED;
+ g_cond_broadcast (&check_cond);
+
+ while (*block_state != UNBLOCK)
+ g_cond_wait (&check_cond, &check_mutex);
+ g_mutex_unlock (&check_mutex);
+}
+
+
+GST_START_TEST (test_play_multithreaded_block_in_describe)
+{
+ GstRTSPConnection *conn;
+ GstRTSPMountPoints *mounts;
+ GstRTSPMediaFactory *factory;
+ gint block_state = BLOCK_ME;
+ GstRTSPMessage *request;
+ GstRTSPMessage *response;
+ GstRTSPStatusCode code;
+ GstRTSPThreadPool *pool;
+
+ pool = gst_rtsp_server_get_thread_pool (server);
+ gst_rtsp_thread_pool_set_max_threads (pool, 2);
+ g_object_unref (pool);
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+ fail_unless (mounts != NULL);
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_launch (factory,
+ "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
+ g_signal_connect (factory, "media-constructed",
+ G_CALLBACK (media_constructed_block), &block_state);
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT "2", factory);
+ g_object_unref (mounts);
+
+ start_server ();
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT "2");
+ iterate ();
+
+ /* do describe, it will not return now as we've blocked it */
+ request = create_request (conn, GST_RTSP_DESCRIBE, NULL);
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+
+ g_mutex_lock (&check_mutex);
+ while (block_state != BLOCKED)
+ g_cond_wait (&check_cond, &check_mutex);
+ g_mutex_unlock (&check_mutex);
+
+ /* Do a second connection while the first one is blocked */
+ do_test_play (NULL);
+
+ /* Now unblock the describe */
+ g_mutex_lock (&check_mutex);
+ block_state = UNBLOCK;
+ g_cond_broadcast (&check_cond);
+ g_mutex_unlock (&check_mutex);
+
+ response = read_response (conn);
+ gst_rtsp_message_parse_response (response, &code, NULL, NULL);
+ fail_unless (code == GST_RTSP_STS_OK);
+ gst_rtsp_message_free (response);
+
+
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+
+}
+
+GST_END_TEST;
+
+
+static void
+new_session_timeout_one (GstRTSPClient * client,
+ GstRTSPSession * session, gpointer user_data)
+{
+ gst_rtsp_session_set_timeout (session, 1);
+
+ g_signal_handlers_disconnect_by_func (client, new_session_timeout_one,
+ user_data);
+}
+
+static void
+session_connected_new_session_cb (GstRTSPServer * server,
+ GstRTSPClient * client, gpointer user_data)
+{
+
+ g_signal_connect (client, "new-session", user_data, NULL);
+}
+
+GST_START_TEST (test_play_multithreaded_timeout_client)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GstRTSPSessionPool *pool;
+ GstRTSPThreadPool *thread_pool;
+
+ thread_pool = gst_rtsp_server_get_thread_pool (server);
+ gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
+ g_object_unref (thread_pool);
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ g_signal_connect (server, "client-connected",
+ G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
+
+ start_server ();
+
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_port);
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup (conn, video_control, &client_port, &session,
+ &video_transport) == GST_RTSP_STS_OK);
+ fail_unless (do_setup (conn, audio_control, &client_port, &session,
+ &audio_transport) == GST_RTSP_STS_OK);
+
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session) == GST_RTSP_STS_OK);
+
+ sleep (7);
+
+ fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 0);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (pool);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_multithreaded_timeout_session)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session1 = NULL;
+ gchar *session2 = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GstRTSPSessionPool *pool;
+ GstRTSPThreadPool *thread_pool;
+
+ thread_pool = gst_rtsp_server_get_thread_pool (server);
+ gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
+ g_object_unref (thread_pool);
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ g_signal_connect (server, "client-connected",
+ G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
+
+ start_server ();
+
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ gst_rtsp_connection_set_remember_session_id (conn, FALSE);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_port);
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup (conn, video_control, &client_port, &session1,
+ &video_transport) == GST_RTSP_STS_OK);
+ fail_unless (do_setup (conn, audio_control, &client_port, &session2,
+ &audio_transport) == GST_RTSP_STS_OK);
+
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 2);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session1) == GST_RTSP_STS_OK);
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session2) == GST_RTSP_STS_OK);
+
+ sleep (7);
+
+ fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
+
+ /* send TEARDOWN request and check that we get 454 Session Not found */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session1) == GST_RTSP_STS_SESSION_NOT_FOUND);
+
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session2) == GST_RTSP_STS_OK);
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (pool);
+ g_free (session1);
+ g_free (session2);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_disconnect)
+{
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ const gchar *audio_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GstRTSPTransport *audio_transport = NULL;
+ GstRTSPSessionPool *pool;
+
+ pool = gst_rtsp_server_get_session_pool (server);
+ g_signal_connect (server, "client-connected",
+ G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
+
+ start_server ();
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+ audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports (&client_port);
+
+ /* do SETUP for video and audio */
+ fail_unless (do_setup (conn, video_control, &client_port, &session,
+ &video_transport) == GST_RTSP_STS_OK);
+ fail_unless (do_setup (conn, audio_control, &client_port, &session,
+ &audio_transport) == GST_RTSP_STS_OK);
+
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session) == GST_RTSP_STS_OK);
+
+ gst_rtsp_connection_free (conn);
+
+ sleep (7);
+
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+ fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
+
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (pool);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_rtsp_transport_free (audio_transport);
+ gst_sdp_message_free (sdp_message);
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+/* Only different with test_play is the specific ports selected */
+
+GST_START_TEST (test_play_specific_server_port)
+{
+ GstRTSPMountPoints *mounts;
+ gchar *service;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAddressPool *pool;
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GSocket *rtp_socket, *rtcp_socket;
+ GSocketAddress *rtp_address, *rtcp_address;
+ guint16 rtp_port, rtcp_port;
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = gst_rtsp_media_factory_new ();
+ pool = gst_rtsp_address_pool_new ();
+ gst_rtsp_address_pool_add_range (pool, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
+ GST_RTSP_ADDRESS_POOL_ANY_IPV4, 7770, 7780, 0);
+ gst_rtsp_media_factory_set_address_pool (factory, pool);
+ g_object_unref (pool);
+ gst_rtsp_media_factory_set_launch (factory, "( " VIDEO_PIPELINE " )");
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+ g_object_unref (mounts);
+
+ /* set port to any */
+ gst_rtsp_server_set_service (server, "0");
+
+ /* attach to default main context */
+ source_id = gst_rtsp_server_attach (server, NULL);
+ fail_if (source_id == 0);
+
+ /* get port */
+ service = gst_rtsp_server_get_service (server);
+ test_port = atoi (service);
+ fail_unless (test_port != 0);
+ g_free (service);
+
+ GST_DEBUG ("rtsp server listening on port %d", test_port);
+
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+ /* get control strings from DESCRIBE response */
+ fail_unless (gst_sdp_message_medias_len (sdp_message) == 1);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
+
+ /* do SETUP for video */
+ fail_unless (do_setup (conn, video_control, &client_port, &session,
+ &video_transport) == GST_RTSP_STS_OK);
+
+ /* send PLAY request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+ session) == GST_RTSP_STS_OK);
+
+ receive_rtp (rtp_socket, &rtp_address);
+ receive_rtcp (rtcp_socket, &rtcp_address, 0);
+
+ fail_unless (G_IS_INET_SOCKET_ADDRESS (rtp_address));
+ fail_unless (G_IS_INET_SOCKET_ADDRESS (rtcp_address));
+ rtp_port =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_address));
+ rtcp_port =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtcp_address));
+ fail_unless (rtp_port >= 7770 && rtp_port <= 7780 && rtp_port % 2 == 0);
+ fail_unless (rtcp_port >= 7770 && rtcp_port <= 7780 && rtcp_port % 2 == 1);
+ fail_unless (rtp_port + 1 == rtcp_port);
+
+ g_object_unref (rtp_address);
+ g_object_unref (rtcp_address);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* FIXME: The rtsp-server always disconnects the transport before
+ * sending the RTCP BYE
+ * receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
+ */
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (rtp_socket);
+ g_object_unref (rtcp_socket);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_smpte_range)
+{
+ start_server ();
+
+ do_test_play ("npt=5-");
+ do_test_play ("smpte=0:00:00-");
+ do_test_play ("smpte=1:00:00-");
+ do_test_play ("smpte=1:00:03-");
+ do_test_play ("clock=20120321T152256Z-");
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+
+static Suite *
+rtspserver_suite (void)
+{
+ Suite *s = suite_create ("rtspserver");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_add_checked_fixture (tc, setup, teardown);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_connect);
+ tcase_add_test (tc, test_describe);
+ tcase_add_test (tc, test_describe_non_existing_mount_point);
+ tcase_add_test (tc, test_setup);
+ tcase_add_test (tc, test_setup_with_require_header);
+ tcase_add_test (tc, test_setup_non_existing_stream);
+ tcase_add_test (tc, test_play);
+ tcase_add_test (tc, test_play_without_session);
+ tcase_add_test (tc, test_bind_already_in_use);
+ tcase_add_test (tc, test_play_multithreaded);
+ tcase_add_test (tc, test_play_multithreaded_block_in_describe);
+ tcase_add_test (tc, test_play_multithreaded_timeout_client);
+ tcase_add_test (tc, test_play_multithreaded_timeout_session);
+ tcase_add_test (tc, test_play_disconnect);
+ tcase_add_test (tc, test_play_specific_server_port);
+ tcase_add_test (tc, test_play_smpte_range);
+ return s;
+}
+
+GST_CHECK_MAIN (rtspserver);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Branko Subasic <branko.subasic@axis.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-media-factory.h>
+#include <rtsp-session-media.h>
+
+#define TEST_PATH "rtsp://localhost:8554/test"
+#define SETUP_URL1 TEST_PATH "/stream=0"
+#define SETUP_URL2 TEST_PATH "/stream=1"
+
+GST_START_TEST (test_setup_url)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url, *setup_url;
+ GstRTSPStream *stream;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPSessionMedia *sm;
+ GstRTSPStreamTransport *trans;
+ GstRTSPTransport *ct;
+ gint match_len;
+ gchar *url_str, *url_str2;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (GST_IS_RTSP_STREAM (stream));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ /* create session media and make sure it matches test path
+ * note that gst_rtsp_session_media_new takes ownership of the media
+ * thus no need to unref it at the bottom of function */
+ sm = gst_rtsp_session_media_new (TEST_PATH, media);
+ fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+ fail_unless (gst_rtsp_session_media_matches (sm, TEST_PATH, &match_len));
+ fail_unless (match_len == strlen (TEST_PATH));
+ fail_unless (gst_rtsp_session_media_get_media (sm) == media);
+
+ /* make a transport for the stream */
+ gst_rtsp_transport_new (&ct);
+ trans = gst_rtsp_session_media_set_transport (sm, stream, ct);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+
+ /* make sure there's no setup url stored initially */
+ fail_unless (gst_rtsp_stream_transport_get_url (trans) == NULL);
+
+ /* now store a setup url and make sure it can be retrieved and that it's correct */
+ fail_unless (gst_rtsp_url_parse (SETUP_URL1, &setup_url) == GST_RTSP_OK);
+ gst_rtsp_stream_transport_set_url (trans, setup_url);
+
+ url_str = gst_rtsp_url_get_request_uri (setup_url);
+ url_str2 =
+ gst_rtsp_url_get_request_uri (gst_rtsp_stream_transport_get_url (trans));
+ fail_if (g_strcmp0 (url_str, url_str2) != 0);
+ g_free (url_str);
+ g_free (url_str2);
+
+ /* check that it's ok to try to store the same url again */
+ gst_rtsp_stream_transport_set_url (trans, setup_url);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ gst_rtsp_url_free (setup_url);
+ gst_rtsp_url_free (url);
+
+ g_object_unref (sm);
+
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_rtsp_state)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPSessionMedia *sm;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (GST_IS_RTSP_STREAM (stream));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ sm = gst_rtsp_session_media_new (TEST_PATH, media);
+ fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+ fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+ GST_RTSP_STATE_INIT);
+
+ gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_READY);
+ fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+ GST_RTSP_STATE_READY);
+
+ gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_SEEKING);
+ fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+ GST_RTSP_STATE_SEEKING);
+
+ gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_PLAYING);
+ fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+ GST_RTSP_STATE_PLAYING);
+
+ gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_RECORDING);
+ fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+ GST_RTSP_STATE_RECORDING);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ gst_rtsp_url_free (url);
+
+ g_object_unref (sm);
+
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_transports)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream1, *stream2;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPSessionMedia *sm;
+ GstRTSPStreamTransport *trans;
+ GstRTSPTransport *ct1, *ct2, *ct3, *ct4;
+ gint match_len;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 audiotestsrc ! rtpgstpay pt=97 name=pay1 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+ stream1 = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (GST_IS_RTSP_STREAM (stream1));
+
+ stream2 = gst_rtsp_media_get_stream (media, 1);
+ fail_unless (GST_IS_RTSP_STREAM (stream2));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ sm = gst_rtsp_session_media_new (TEST_PATH, media);
+ fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+ fail_unless (gst_rtsp_session_media_matches (sm, TEST_PATH, &match_len));
+ fail_unless (match_len == strlen (TEST_PATH));
+
+ gst_rtsp_transport_new (&ct1);
+ trans = gst_rtsp_session_media_set_transport (sm, stream1, ct1);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+
+ gst_rtsp_transport_new (&ct2);
+ trans = gst_rtsp_session_media_set_transport (sm, stream1, ct2);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+
+ gst_rtsp_transport_new (&ct3);
+ trans = gst_rtsp_session_media_set_transport (sm, stream2, ct3);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 1) == trans);
+
+ gst_rtsp_transport_new (&ct4);
+ trans = gst_rtsp_session_media_set_transport (sm, stream2, ct4);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 1) == trans);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ gst_rtsp_url_free (url);
+
+ g_object_unref (sm);
+
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_time_and_rtpinfo)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream1, *stream2;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPSessionMedia *sm;
+ GstClockTime base_time;
+ gchar *rtpinfo;
+ GstRTSPTransport *ct1;
+ GstRTSPStreamTransport *trans;
+ GstRTSPUrl *setup_url;
+ gchar **streaminfo;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc do-timestamp=true timestamp-offset=0 ! rtpvrawpay pt=96 name=pay0 "
+ "audiotestsrc do-timestamp=true timestamp-offset=1000000000 ! rtpgstpay pt=97 name=pay1 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+ stream1 = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (GST_IS_RTSP_STREAM (stream1));
+
+ stream2 = gst_rtsp_media_get_stream (media, 1);
+ fail_unless (GST_IS_RTSP_STREAM (stream2));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ sm = gst_rtsp_session_media_new (TEST_PATH, media);
+ fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+
+ base_time = gst_rtsp_session_media_get_base_time (sm);
+ fail_unless_equals_int64 (base_time, 0);
+
+ rtpinfo = gst_rtsp_session_media_get_rtpinfo (sm);
+ fail_unless (rtpinfo == NULL);
+
+ gst_rtsp_transport_new (&ct1);
+ trans = gst_rtsp_session_media_set_transport (sm, stream1, ct1);
+ fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+ fail_unless (gst_rtsp_url_parse (SETUP_URL1, &setup_url) == GST_RTSP_OK);
+ gst_rtsp_stream_transport_set_url (trans, setup_url);
+
+ base_time = gst_rtsp_session_media_get_base_time (sm);
+ fail_unless_equals_int64 (base_time, 0);
+
+ rtpinfo = gst_rtsp_session_media_get_rtpinfo (sm);
+ streaminfo = g_strsplit (rtpinfo, ",", 1);
+ g_free (rtpinfo);
+
+ fail_unless (g_strstr_len (streaminfo[0], -1, "url=") != NULL);
+ fail_unless (g_strstr_len (streaminfo[0], -1, "seq=") != NULL);
+ fail_unless (g_strstr_len (streaminfo[0], -1, "rtptime=") != NULL);
+ fail_unless (g_strstr_len (streaminfo[0], -1, SETUP_URL1) != NULL);
+
+ g_strfreev (streaminfo);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ rtpinfo = gst_rtsp_session_media_get_rtpinfo (sm);
+ fail_unless (rtpinfo == NULL);
+
+ gst_rtsp_url_free (setup_url);
+ gst_rtsp_url_free (url);
+
+ g_object_unref (sm);
+
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_allocate_channels)
+{
+ GstRTSPMediaFactory *factory;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream;
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+ GstRTSPSessionMedia *sm;
+ GstRTSPRange range;
+
+ factory = gst_rtsp_media_factory_new ();
+ fail_if (gst_rtsp_media_factory_is_shared (factory));
+ fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (GST_IS_RTSP_STREAM (stream));
+
+ pool = gst_rtsp_thread_pool_new ();
+ thread = gst_rtsp_thread_pool_get_thread (pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+ fail_unless (gst_rtsp_media_prepare (media, thread));
+
+ sm = gst_rtsp_session_media_new (TEST_PATH, media);
+ fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+
+ fail_unless (gst_rtsp_session_media_alloc_channels (sm, &range));
+ fail_unless_equals_int (range.min, 0);
+ fail_unless_equals_int (range.max, 1);
+
+ fail_unless (gst_rtsp_session_media_alloc_channels (sm, &range));
+ fail_unless_equals_int (range.min, 2);
+ fail_unless_equals_int (range.max, 3);
+
+ fail_unless (gst_rtsp_media_unprepare (media));
+
+ gst_rtsp_url_free (url);
+
+ g_object_unref (sm);
+
+ g_object_unref (factory);
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+static Suite *
+rtspsessionmedia_suite (void)
+{
+ Suite *s = suite_create ("rtspsessionmedia");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_setup_url);
+ tcase_add_test (tc, test_rtsp_state);
+ tcase_add_test (tc, test_transports);
+ tcase_add_test (tc, test_time_and_rtpinfo);
+ tcase_add_test (tc, test_allocate_channels);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspsessionmedia);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2014 Sebastian Rasmussen <sebras@hotmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <rtsp-session-pool.h>
+
+typedef struct
+{
+ GstRTSPSession *sessions[3];
+ GstRTSPFilterResult response[3];
+} Responses;
+
+static GstRTSPFilterResult
+filter_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
+ gpointer user_data)
+{
+ Responses *responses = (Responses *) user_data;
+ gint i;
+
+ for (i = 0; i < 3; i++)
+ if (session == responses->sessions[i])
+ return responses->response[i];
+
+ return GST_RTSP_FILTER_KEEP;
+}
+
+GST_START_TEST (test_pool)
+{
+ GstRTSPSessionPool *pool;
+ GstRTSPSession *session1, *session2, *session3;
+ GstRTSPSession *compare;
+ gchar *session1id, *session2id, *session3id;
+ GList *list;
+ guint maxsessions;
+ GSource *source;
+ guint sourceid;
+
+ pool = gst_rtsp_session_pool_new ();
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 0);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 0);
+
+ gst_rtsp_session_pool_set_max_sessions (pool, 3);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+
+ session1 = gst_rtsp_session_pool_create (pool);
+ fail_unless (GST_IS_RTSP_SESSION (session1));
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 1);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+ session1id = g_strdup (gst_rtsp_session_get_sessionid (session1));
+
+ session2 = gst_rtsp_session_pool_create (pool);
+ fail_unless (GST_IS_RTSP_SESSION (session2));
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 2);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+ session2id = g_strdup (gst_rtsp_session_get_sessionid (session2));
+
+ session3 = gst_rtsp_session_pool_create (pool);
+ fail_unless (GST_IS_RTSP_SESSION (session3));
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 3);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+ session3id = g_strdup (gst_rtsp_session_get_sessionid (session3));
+
+ fail_if (GST_IS_RTSP_SESSION (gst_rtsp_session_pool_create (pool)));
+
+ compare = gst_rtsp_session_pool_find (pool, session1id);
+ fail_unless (compare == session1);
+ g_object_unref (compare);
+ compare = gst_rtsp_session_pool_find (pool, session2id);
+ fail_unless (compare == session2);
+ g_object_unref (compare);
+ compare = gst_rtsp_session_pool_find (pool, session3id);
+ fail_unless (compare == session3);
+ g_object_unref (compare);
+ fail_unless (gst_rtsp_session_pool_find (pool, "") == NULL);
+
+ fail_unless (gst_rtsp_session_pool_remove (pool, session2));
+ g_object_unref (session2);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 2);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+
+ gst_rtsp_session_pool_set_max_sessions (pool, 2);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 2);
+ fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 2);
+
+ session2 = gst_rtsp_session_pool_create (pool);
+ fail_if (GST_IS_RTSP_SESSION (session2));
+
+ {
+ list = gst_rtsp_session_pool_filter (pool, NULL, NULL);
+ fail_unless_equals_int (g_list_length (list), 2);
+ fail_unless (g_list_find (list, session1) != NULL);
+ fail_unless (g_list_find (list, session3) != NULL);
+ g_list_free_full (list, (GDestroyNotify) g_object_unref);
+ }
+
+ {
+ Responses responses = {
+ {session1, session2, session3},
+ {GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP},
+ };
+
+ list = gst_rtsp_session_pool_filter (pool, filter_func, &responses);
+ fail_unless (list == NULL);
+ }
+
+ {
+ Responses responses = {
+ {session1, session2, session3},
+ {GST_RTSP_FILTER_REF, GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP},
+ };
+
+ list = gst_rtsp_session_pool_filter (pool, filter_func, &responses);
+ fail_unless_equals_int (g_list_length (list), 1);
+ fail_unless (g_list_nth_data (list, 0) == session1);
+ g_list_free_full (list, (GDestroyNotify) g_object_unref);
+ }
+
+ {
+ Responses responses = {
+ {session1, session2, session3},
+ {GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_REMOVE},
+ };
+
+ list = gst_rtsp_session_pool_filter (pool, filter_func, &responses);
+ fail_unless_equals_int (g_list_length (list), 0);
+ g_list_free (list);
+ }
+
+ compare = gst_rtsp_session_pool_find (pool, session1id);
+ fail_unless (compare == session1);
+ g_object_unref (compare);
+ fail_unless (gst_rtsp_session_pool_find (pool, session2id) == NULL);
+ fail_unless (gst_rtsp_session_pool_find (pool, session3id) == NULL);
+
+ g_object_get (pool, "max-sessions", &maxsessions, NULL);
+ fail_unless_equals_int (maxsessions, 2);
+
+ g_object_set (pool, "max-sessions", 3, NULL);
+ g_object_get (pool, "max-sessions", &maxsessions, NULL);
+ fail_unless_equals_int (maxsessions, 3);
+
+ fail_unless_equals_int (gst_rtsp_session_pool_cleanup (pool), 0);
+
+ gst_rtsp_session_set_timeout (session1, 1);
+
+ source = gst_rtsp_session_pool_create_watch (pool);
+ fail_unless (source != NULL);
+
+ sourceid = g_source_attach (source, NULL);
+ fail_unless (sourceid != 0);
+
+ while (!g_main_context_iteration (NULL, TRUE));
+
+ g_source_unref (source);
+
+ g_object_unref (session1);
+ g_object_unref (session3);
+
+ g_free (session1id);
+ g_free (session2id);
+ g_free (session3id);
+
+ g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspsessionpool_suite (void)
+{
+ Suite *s = suite_create ("rtspsessionpool");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 15);
+ tcase_add_test (tc, test_pool);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspsessionpool);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Axis Communications AB <dev-gstreamer at axis dot com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-stream.h>
+#include <rtsp-address-pool.h>
+
+GST_START_TEST (test_get_sockets)
+{
+ GstPad *srcpad;
+ GstElement *pay;
+ GstRTSPStream *stream;
+ GstBin *bin;
+ GstElement *rtpbin;
+ GSocket *socket;
+ gboolean have_ipv4;
+ gboolean have_ipv6;
+
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ gst_pad_set_active (srcpad, TRUE);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+ rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+ fail_unless (rtpbin != NULL);
+ bin = GST_BIN (gst_bin_new ("testbin"));
+ fail_unless (bin != NULL);
+ fail_unless (gst_bin_add (bin, rtpbin));
+
+ fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+ socket = gst_rtsp_stream_get_rtp_socket (stream, G_SOCKET_FAMILY_IPV4);
+ have_ipv4 = (socket != NULL);
+ if (have_ipv4) {
+ fail_unless (g_socket_get_fd (socket) >= 0);
+ g_object_unref (socket);
+ }
+
+ socket = gst_rtsp_stream_get_rtcp_socket (stream, G_SOCKET_FAMILY_IPV4);
+ if (have_ipv4) {
+ fail_unless (socket != NULL);
+ fail_unless (g_socket_get_fd (socket) >= 0);
+ g_object_unref (socket);
+ } else {
+ fail_unless (socket == NULL);
+ }
+
+ socket = gst_rtsp_stream_get_rtp_socket (stream, G_SOCKET_FAMILY_IPV6);
+ have_ipv6 = (socket != NULL);
+ if (have_ipv6) {
+ fail_unless (g_socket_get_fd (socket) >= 0);
+ g_object_unref (socket);
+ }
+
+ socket = gst_rtsp_stream_get_rtcp_socket (stream, G_SOCKET_FAMILY_IPV6);
+ if (have_ipv6) {
+ fail_unless (socket != NULL);
+ fail_unless (g_socket_get_fd (socket) >= 0);
+ g_object_unref (socket);
+ } else {
+ fail_unless (socket == NULL);
+ }
+
+ /* check that at least one family is available */
+ fail_unless (have_ipv4 || have_ipv6);
+
+ fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+
+ gst_object_unref (bin);
+ gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_get_multicast_address)
+{
+ GstPad *srcpad;
+ GstElement *pay;
+ GstRTSPStream *stream;
+ GstRTSPAddressPool *pool;
+ GstRTSPAddress *addr1;
+ GstRTSPAddress *addr2;
+
+ srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+ fail_unless (srcpad != NULL);
+ gst_pad_set_active (srcpad, TRUE);
+ pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+ fail_unless (pay != NULL);
+ stream = gst_rtsp_stream_new (0, pay, srcpad);
+ fail_unless (stream != NULL);
+ gst_object_unref (pay);
+ gst_object_unref (srcpad);
+
+ pool = gst_rtsp_address_pool_new ();
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "233.252.0.0", "233.252.0.0", 5000, 5001, 1));
+ fail_unless (gst_rtsp_address_pool_add_range (pool,
+ "FF11:DB8::1", "FF11:DB8::1", 5002, 5003, 1));
+ gst_rtsp_stream_set_address_pool (stream, pool);
+
+ addr1 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+ fail_unless (addr1 != NULL);
+ fail_unless_equals_string (addr1->address, "233.252.0.0");
+ fail_unless_equals_int (addr1->port, 5000);
+ fail_unless_equals_int (addr1->n_ports, 2);
+
+ addr2 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+ fail_unless (addr2 != NULL);
+ fail_unless_equals_string (addr2->address, "233.252.0.0");
+ fail_unless_equals_int (addr2->port, 5000);
+ fail_unless_equals_int (addr2->n_ports, 2);
+
+ gst_rtsp_address_free (addr1);
+ gst_rtsp_address_free (addr2);
+
+ addr1 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
+ fail_unless (addr1 != NULL);
+ fail_unless (!g_ascii_strcasecmp (addr1->address, "FF11:DB8::1"));
+ fail_unless_equals_int (addr1->port, 5002);
+ fail_unless_equals_int (addr1->n_ports, 2);
+
+ addr2 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
+ fail_unless (addr2 != NULL);
+ fail_unless (!g_ascii_strcasecmp (addr2->address, "FF11:DB8::1"));
+ fail_unless_equals_int (addr2->port, 5002);
+ fail_unless_equals_int (addr2->n_ports, 2);
+
+ gst_rtsp_address_free (addr1);
+ gst_rtsp_address_free (addr2);
+
+ g_object_unref (pool);
+
+ gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspstream_suite (void)
+{
+ Suite *s = suite_create ("rtspstream");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_add_test (tc, test_get_sockets);
+ tcase_add_test (tc, test_get_multicast_address);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspstream);
--- /dev/null
+/* GStreamer
+ * unit tests for GstRTSPThreadPool
+ * Copyright (C) 2013 Axis Communications <dev-gstreamer at axis dot com>
+ * @author Ognyan Tonchev <ognyan at axis dot com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-thread-pool.h>
+
+GST_START_TEST (test_pool_get_thread)
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ pool = gst_rtsp_thread_pool_new ();
+ fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+ thread = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread));
+ /* one ref is hold by the pool */
+ fail_unless_equals_int (GST_MINI_OBJECT_REFCOUNT (thread), 2);
+
+ gst_rtsp_thread_stop (thread);
+ g_object_unref (pool);
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_get_media_thread)
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread;
+
+ pool = gst_rtsp_thread_pool_new ();
+ fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+ thread = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_MEDIA,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread));
+ /* one ref is hold by the pool */
+ fail_unless_equals_int (GST_MINI_OBJECT_REFCOUNT (thread), 2);
+
+ gst_rtsp_thread_stop (thread);
+ g_object_unref (pool);
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_get_thread_reuse)
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread1;
+ GstRTSPThread *thread2;
+
+ pool = gst_rtsp_thread_pool_new ();
+ fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+ gst_rtsp_thread_pool_set_max_threads (pool, 1);
+
+ thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread1));
+
+ thread2 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread2));
+
+ fail_unless (thread2 == thread1);
+ /* one ref is hold by the pool */
+ fail_unless_equals_int (GST_MINI_OBJECT_REFCOUNT (thread1), 3);
+
+ gst_rtsp_thread_stop (thread1);
+ gst_rtsp_thread_stop (thread2);
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static void
+do_test_pool_max_thread (gboolean use_property)
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread1;
+ GstRTSPThread *thread2;
+ GstRTSPThread *thread3;
+ gint max_threads;
+
+ pool = gst_rtsp_thread_pool_new ();
+ fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+ if (use_property) {
+ g_object_get (pool, "max-threads", &max_threads, NULL);
+ fail_unless_equals_int (max_threads, 1);
+ } else {
+ fail_unless_equals_int (gst_rtsp_thread_pool_get_max_threads (pool), 1);
+ }
+
+ thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread1));
+
+ thread2 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread2));
+
+ fail_unless (thread1 == thread2);
+
+ gst_rtsp_thread_stop (thread1);
+ gst_rtsp_thread_stop (thread2);
+
+ if (use_property) {
+ g_object_set (pool, "max-threads", 2, NULL);
+ g_object_get (pool, "max-threads", &max_threads, NULL);
+ fail_unless_equals_int (max_threads, 2);
+ } else {
+ gst_rtsp_thread_pool_set_max_threads (pool, 2);
+ fail_unless_equals_int (gst_rtsp_thread_pool_get_max_threads (pool), 2);
+ }
+
+ thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread1));
+
+ thread2 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread2));
+
+ thread3 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread3));
+
+ fail_unless (thread2 != thread1);
+ fail_unless (thread3 == thread2 || thread3 == thread1);
+
+ gst_rtsp_thread_stop (thread1);
+ gst_rtsp_thread_stop (thread2);
+ gst_rtsp_thread_stop (thread3);
+
+ if (use_property) {
+ g_object_set (pool, "max-threads", 0, NULL);
+ g_object_get (pool, "max-threads", &max_threads, NULL);
+ fail_unless_equals_int (max_threads, 0);
+ } else {
+ gst_rtsp_thread_pool_set_max_threads (pool, 0);
+ fail_unless_equals_int (gst_rtsp_thread_pool_get_max_threads (pool), 0);
+ }
+
+ thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_if (GST_IS_RTSP_THREAD (thread1));
+
+ g_object_unref (pool);
+
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_START_TEST (test_pool_max_threads)
+{
+ do_test_pool_max_thread (FALSE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_max_threads_property)
+{
+ do_test_pool_max_thread (TRUE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_thread_copy)
+{
+ GstRTSPThreadPool *pool;
+ GstRTSPThread *thread1;
+ GstRTSPThread *thread2;
+
+ pool = gst_rtsp_thread_pool_new ();
+ fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+ thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+ NULL);
+ fail_unless (GST_IS_RTSP_THREAD (thread1));
+ fail_unless (GST_IS_MINI_OBJECT_TYPE (thread1, GST_TYPE_RTSP_THREAD));
+
+ thread2 = GST_RTSP_THREAD (gst_mini_object_copy (GST_MINI_OBJECT (thread1)));
+ fail_unless (GST_IS_RTSP_THREAD (thread2));
+ fail_unless (GST_IS_MINI_OBJECT_TYPE (thread2, GST_TYPE_RTSP_THREAD));
+
+ gst_rtsp_thread_stop (thread1);
+ gst_rtsp_thread_stop (thread2);
+ g_object_unref (pool);
+ gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspthreadpool_suite (void)
+{
+ Suite *s = suite_create ("rtspthreadpool");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_pool_get_thread);
+ tcase_add_test (tc, test_pool_get_media_thread);
+ tcase_add_test (tc, test_pool_get_thread_reuse);
+ tcase_add_test (tc, test_pool_max_threads);
+ tcase_add_test (tc, test_pool_max_threads_property);
+ tcase_add_test (tc, test_pool_thread_copy);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtspthreadpool);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2013 Sebastian Rasmussen <sebras@hotmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-token.h>
+
+GST_START_TEST (test_token)
+{
+ GstRTSPToken *token;
+ GstRTSPToken *token2;
+ GstRTSPToken *copy;
+ GstStructure *str;
+
+ token = gst_rtsp_token_new_empty ();
+ fail_if (gst_rtsp_token_is_allowed (token, "missing"));
+ gst_rtsp_token_unref (token);
+
+ token = gst_rtsp_token_new ("role", G_TYPE_STRING, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+ fail_unless (gst_rtsp_token_is_allowed (token, "permission1"));
+ fail_if (gst_rtsp_token_is_allowed (token, "permission2"));
+ fail_if (gst_rtsp_token_is_allowed (token, "missing"));
+ copy = GST_RTSP_TOKEN (gst_mini_object_copy (GST_MINI_OBJECT (token)));
+ gst_rtsp_token_unref (token);
+ fail_unless_equals_string (gst_rtsp_token_get_string (copy, "role"), "user");
+ fail_unless (gst_rtsp_token_is_allowed (copy, "permission1"));
+ fail_if (gst_rtsp_token_is_allowed (copy, "permission2"));
+ fail_if (gst_rtsp_token_is_allowed (copy, "missing"));
+ gst_rtsp_token_unref (copy);
+
+ token = gst_rtsp_token_new ("role", G_TYPE_STRING, "user",
+ "permission1", G_TYPE_BOOLEAN, TRUE,
+ "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+ fail_unless (gst_rtsp_token_is_allowed (token, "permission1"));
+ fail_if (gst_rtsp_token_is_allowed (token, "permission2"));
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+
+ fail_unless (gst_mini_object_is_writable (GST_MINI_OBJECT (token)));
+ fail_unless (gst_rtsp_token_writable_structure (token) != NULL);
+ fail_unless (gst_rtsp_token_get_structure (token) != NULL);
+
+ token2 = gst_rtsp_token_ref (token);
+
+ fail_if (gst_mini_object_is_writable (GST_MINI_OBJECT (token)));
+ ASSERT_CRITICAL (fail_unless (gst_rtsp_token_writable_structure (token) ==
+ NULL));
+ fail_unless (gst_rtsp_token_get_structure (token) != NULL);
+
+ gst_rtsp_token_unref (token2);
+
+ fail_unless (gst_mini_object_is_writable (GST_MINI_OBJECT (token)));
+ fail_unless (gst_rtsp_token_writable_structure (token) != NULL);
+ fail_unless (gst_rtsp_token_get_structure (token) != NULL);
+
+ str = gst_rtsp_token_writable_structure (token);
+ gst_structure_set (str, "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+ fail_unless (gst_rtsp_token_is_allowed (token, "permission1"));
+ fail_unless (gst_rtsp_token_is_allowed (token, "permission2"));
+ fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+ gst_rtsp_token_unref (token);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtsptoken_suite (void)
+{
+ Suite *s = suite_create ("rtsptoken");
+ TCase *tc = tcase_create ("general");
+
+ suite_add_tcase (s, tc);
+ tcase_set_timeout (tc, 20);
+ tcase_add_test (tc, test_token);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtsptoken);
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+static gboolean
+timeout (GMainLoop * loop, gboolean ignored)
+{
+ g_main_loop_quit (loop);
+ return FALSE;
+}
+
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+ GstRTSPServer *server;
+ guint id;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* attach the server to the default maincontext */
+ if ((id = gst_rtsp_server_attach (server, NULL)) == 0)
+ goto failed;
+
+ g_timeout_add_seconds (2, (GSourceFunc) timeout, loop);
+
+ /* start serving */
+ g_main_loop_run (loop);
+
+ /* cleanup */
+ g_source_remove (id);
+ g_object_unref (server);
+ g_main_loop_unref (loop);
+
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+#define TIMEOUT 2
+
+static gboolean timeout_1 (GMainLoop * loop, gboolean ignored);
+
+static guint id;
+static gint rounds = 3;
+static GstRTSPServer *server;
+
+static gboolean
+timeout_2 (GMainLoop * loop, gboolean ignored)
+{
+ rounds--;
+ if (rounds > 0) {
+ id = gst_rtsp_server_attach (server, NULL);
+ g_print ("have attached\n");
+ g_timeout_add_seconds (TIMEOUT, (GSourceFunc) timeout_1, loop);
+ } else {
+ g_main_loop_quit (loop);
+ }
+ return FALSE;
+}
+
+static gboolean
+timeout_1 (GMainLoop * loop, gboolean ignored)
+{
+ g_source_remove (id);
+ g_print ("have removed\n");
+ g_timeout_add_seconds (TIMEOUT, (GSourceFunc) timeout_2, loop);
+ return FALSE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GMainLoop *loop;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* create a server instance */
+ server = gst_rtsp_server_new ();
+
+ /* attach the server to the default maincontext */
+ if ((id = gst_rtsp_server_attach (server, NULL)) == 0)
+ goto failed;
+ g_print ("have attached\n");
+
+ g_timeout_add_seconds (TIMEOUT, (GSourceFunc) timeout_1, loop);
+
+ /* start serving */
+ g_main_loop_run (loop);
+
+ /* cleanup */
+ g_object_unref (server);
+ g_main_loop_unref (loop);
+
+ g_print ("quit\n");
+ return 0;
+
+ /* ERRORS */
+failed:
+ {
+ g_print ("failed to attach the server\n");
+ return -1;
+ }
+}
projects / framework / multimedia / gst-rtsp-server.git / commitdiff
